This patch tracks total number of bytes acked for a TCP socket.
This is the sum of all changes done to tp->snd_una, and allows
for precise tracking of delivered data.

RFC4898 named this : tcpEStatsAppHCThruOctetsAcked

This is a 64bit field, and can be fetched both from TCP_INFO
getsockopt() if one has a handle on a TCP socket, or from inet_diag
netlink facility (iproute2/ss patch will follow)

Note that tp->bytes_acked was placed near tp->snd_una for
best data locality and minimal performance impact.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Cc: Matt Mathis <mattmathis@google.com>
Cc: Eric Salo <salo@google.com>
Cc: Martin Lau <kafai@fb.com>
Cc: Chris Rapier <rapier@psc.edu>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Ensure that we either see that the buffer has write space
in tcp_poll() or that we perform a wakeup from the input
side. Did not run into any actual problem here, but thought
that we should make things explicit.
Signed-off-by: NJason Baron <jbaron@akamai.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Since retransmitted segments are not used for RTT estimation, previously
SACKed segments present in the rtx queue are used. This estimation can be
several times larger than the actual RTT. When a cumulative ack covers both
previously SACKed and retransmitted segments, CC may thus get a bogus RTT.

Such segments previously had an RTT estimation in tcp_sacktag_one(), so it
seems reasonable to not reuse them in tcp_clean_rtx_queue() at all.

Afaik, this has had no effect on SRTT/RTO because of Karn's check.
Signed-off-by: NKenneth Klette Jonassen <kennetkl@ifi.uio.no>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Tested-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Fast Open has been using an experimental option with a magic number
(RFC6994). This patch makes the client by default use the RFC7413
option (34) to get and send Fast Open cookies.  This patch makes
the client solicit cookies from a given server first with the
RFC7413 option. If that fails to elicit a cookie, then it tries
the RFC6994 experimental option. If that also fails, it uses the
RFC7413 option on all subsequent connect attempts.  If the server
returns a Fast Open cookie then the client caches the form of the
option that successfully elicited a cookie, and uses that form on
later connects when it presents that cookie.

The idea is to gradually obsolete the use of experimental options as
the servers and clients upgrade, while keeping the interoperability
meanwhile.
Signed-off-by: NDaniel Lee <Longinus00@gmail.com>
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>

Fast Open has been using the experimental option with a magic number
(RFC6994) to request and grant Fast Open cookies. This patch enables
the server to support the official IANA option 34 in RFC7413 in
addition.

The change has passed all existing Fast Open tests with both
old and new options at Google.
Signed-off-by: NDaniel Lee <Longinus00@gmail.com>
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>

The ipv4 code uses a mixture of coding styles. In some instances check
for non-NULL pointer is done as x != NULL and sometimes as x. x is
preferred according to checkpatch and this patch makes the code
consistent by adopting the latter form.

No changes detected by objdiff.
Signed-off-by: NIan Morris <ipm@chirality.org.uk>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The ipv4 code uses a mixture of coding styles. In some instances check
for NULL pointer is done as x == NULL and sometimes as !x. !x is
preferred according to checkpatch and this patch makes the code
consistent by adopting the latter form.

No changes detected by objdiff.
Signed-off-by: NIan Morris <ipm@chirality.org.uk>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

On processing cumulative ACKs, the FRTO code was not checking the
SACKed bit, meaning that there could be a spurious FRTO undo on a
cumulative ACK of a previously SACKed skb.

The FRTO code should only consider a cumulative ACK to indicate that
an original/unretransmitted skb is newly ACKed if the skb was not yet
SACKed.

The effect of the spurious FRTO undo would typically be to make the
connection think that all previously-sent packets were in flight when
they really weren't, leading to a stall and an RTO.
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>

After commit 1fb6f159 ("tcp: add tcp_conn_request"),
tcp_syn_flood_action() is no longer used from IPv6.

We can make it static, by moving it above tcp_conn_request()
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reviewed-by: NOctavian Purdila <octavian.purdila@intel.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

ss should display ipv4 mapped request sockets like this :

tcp    SYN-RECV   0      0  ::ffff:192.168.0.1:8080   ::ffff:192.0.2.1:35261

and not like this :

tcp    SYN-RECV   0      0  192.168.0.1:8080   192.0.2.1:35261

We should init ireq->ireq_family based on listener sk_family,
not the actual protocol carried by SYN packet.

This means we can set ireq_family in inet_reqsk_alloc()

Fixes: 3f66b083 ("inet: introduce ireq_family")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

When request sock are put in ehash table, the whole notion
of having a previous request to update dl_next is pointless.

Also, following patch will get rid of big purge timer,
so we want to delete a request sock without holding listener lock.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

While testing last patch series, I found req sock refcounting was wrong.

We must set skc_refcnt to 1 for all request socks added in hashes,
but also on request sockets created by FastOpen or syncookies.

It is tricky because we need to defer this initialization so that
future RCU lookups do not try to take a refcount on a not yet
fully initialized request socket.

Also get rid of ireq_refcnt alias.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Fixes: 13854e5a ("inet: add proper refcounting to request sock")
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The listener field in struct tcp_request_sock is a pointer
back to the listener. We now have req->rsk_listener, so TCP
only needs one boolean and not a full pointer.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Once we'll be able to lookup request sockets in ehash table,
we'll need to get access to listener which created this request.

This avoid doing a lookup to find the listener, which benefits
for a more solid SO_REUSEPORT, and is needed once we no
longer queue request sock into a listener private queue.

Note that 'struct tcp_request_sock'->listener could be reduced
to a single bit, as TFO listener should match req->rsk_listener.
TFO will no longer need to hold a reference on the listener.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

inet_reqsk_alloc() is becoming fat and should not be inlined.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

listener socket can be used to set net pointer, and will
be later used to hold a reference on listener.

Add a const qualifier to first argument (struct request_sock_ops *),
and factorize all write_pnet(&ireq->ireq_net, sock_net(sk));
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_oow_rate_limited() is hardly used in fast path, there is
no point inlining it.
Signed-of-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This big helper is called once from tcp_conn_request(), there is no
point having it in an include. Compiler will inline it anyway.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Once request socks will be in ehash table, they will need to have
a valid ir_iff field.

This is currently true only for IPv6. This patch extends support
for IPv4 as well.

This means inet_diag_fill_req() can now properly use ir_iif,
which is better for IPv6 link locals anyway, as request sockets
and established sockets will propagate consistent netlink idiag_if.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

I forgot to update dccp_v6_conn_request() & cookie_v6_check().
They both need to set ireq->ireq_net and ireq->ir_cookie

Lets clear ireq->ir_cookie in inet_reqsk_alloc()
Signed-off-by: NEric Dumazet <edumazet@google.com>
Fixes: 33cf7c90 ("net: add real socket cookies")
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

I forgot to use write_pnet() in three locations.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Fixes: 33cf7c90 ("net: add real socket cookies")
Reported-by: Nkbuild test robot <fengguang.wu@intel.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

A long standing problem in netlink socket dumps is the use
of kernel socket addresses as cookies.

1) It is a security concern.

2) Sockets can be reused quite quickly, so there is
   no guarantee a cookie is used once and identify
   a flow.

3) request sock, establish sock, and timewait socks
   for a given flow have different cookies.

Part of our effort to bring better TCP statistics requires
to switch to a different allocator.

In this patch, I chose to use a per network namespace 64bit generator,
and to use it only in the case a socket needs to be dumped to netlink.
(This might be refined later if needed)

Note that I tried to carry cookies from request sock, to establish sock,
then timewait sockets.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Eric Salo <salo@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_should_expand_sndbuf() does not expand the send buffer if we have
filled the congestion window.

However, it should use tcp_packets_in_flight() instead of
tp->packets_out to make this check.

Testing has established that the difference matters a lot if there are
many SACKed packets, causing a needless performance shortfall.
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Ensure that in state ESTABLISHED, where the connection is represented
by a tcp_sock, we rate limit dupacks in response to incoming packets
(a) with TCP timestamps that fail PAWS checks, or (b) with sequence
numbers or ACK numbers that are out of the acceptable window.

We do not send a dupack in response to out-of-window packets if it has
been less than sysctl_tcp_invalid_ratelimit (default 500ms) since we
last sent a dupack in response to an out-of-window packet.

There is already a similar (although global) rate-limiting mechanism
for "challenge ACKs". When deciding whether to send a challence ACK,
we first consult the new per-connection rate limit, and then the
global rate limit.
Reported-by: NAvery Fay <avery@mixpanel.com>
Signed-off-by: NNeal Cardwell <ncardwell@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>

Helpers for mitigating ACK loops by rate-limiting dupacks sent in
response to incoming out-of-window packets.

This patch includes:

- rate-limiting logic
- sysctl to control how often we allow dupacks to out-of-window packets
- SNMP counter for cases where we rate-limited our dupack sending

The rate-limiting logic in this patch decides to not send dupacks in
response to out-of-window segments if (a) they are SYNs or pure ACKs
and (b) the remote endpoint is sending them faster than the configured
rate limit.

We rate-limit our responses rather than blocking them entirely or
resetting the connection, because legitimate connections can rely on
dupacks in response to some out-of-window segments. For example, zero
window probes are typically sent with a sequence number that is below
the current window, and ZWPs thus expect to thus elicit a dupack in
response.

We allow dupacks in response to TCP segments with data, because these
may be spurious retransmissions for which the remote endpoint wants to
receive DSACKs. This is safe because segments with data can't
realistically be part of ACK loops, which by their nature consist of
each side sending pure/data-less ACKs to each other.

The dupack interval is controlled by a new sysctl knob,
tcp_invalid_ratelimit, given in milliseconds, in case an administrator
needs to dial this upward in the face of a high-rate DoS attack. The
name and units are chosen to be analogous to the existing analogous
knob for ICMP, icmp_ratelimit.

The default value for tcp_invalid_ratelimit is 500ms, which allows at
most one such dupack per 500ms. This is chosen to be 2x faster than
the 1-second minimum RTO interval allowed by RFC 6298 (section 2, rule
2.4). We allow the extra 2x factor because network delay variations
can cause packets sent at 1 second intervals to be compressed and
arrive much closer.
Reported-by: NAvery Fay <avery@mixpanel.com>
Signed-off-by: NNeal Cardwell <ncardwell@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>

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>