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  1. 12 11月, 2014 1 次提交
    • J
      net: Convert LIMIT_NETDEBUG to net_dbg_ratelimited · ba7a46f1
      由 Joe Perches 提交于 
      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>
      ba7a46f1
  3. 06 11月, 2014 1 次提交
    • M
      tcp: zero retrans_stamp if all retrans were acked · 1f37bf87
      由 Marcelo Leitner 提交于 
      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>
      1f37bf87
  5. 05 11月, 2014 1 次提交
    • F
      net: allow setting ecn via routing table · f7b3bec6
      由 Florian Westphal 提交于 
      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>
      f7b3bec6
  7. 31 10月, 2014 2 次提交
  9. 30 10月, 2014 1 次提交
    • E
      tcp: allow for bigger reordering level · dca145ff
      由 Eric Dumazet 提交于 
      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>
      dca145ff
  11. 15 10月, 2014 1 次提交
    • E
      tcp: fix tcp_ack() performance problem · ad971f61
      由 Eric Dumazet 提交于 
      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>
      ad971f61
  13. 30 9月, 2014 2 次提交
  15. 29 9月, 2014 5 次提交
  17. 28 9月, 2014 2 次提交
  19. 24 9月, 2014 1 次提交
  21. 23 9月, 2014 1 次提交
  23. 20 9月, 2014 1 次提交
  25. 16 9月, 2014 3 次提交
  27. 06 9月, 2014 2 次提交
  29. 23 8月, 2014 1 次提交
    • Y
      tcp: improve undo on timeout · 989e04c5
      由 Yuchung Cheng 提交于 
      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>
      989e04c5
  31. 15 8月, 2014 2 次提交
    • N
      tcp: fix ssthresh and undo for consecutive short FRTO episodes · 0c9ab092
      由 Neal Cardwell 提交于 
      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>
      0c9ab092
    • H
      tcp: don't allow syn packets without timestamps to pass tcp_tw_recycle logic · a26552af
      由 Hannes Frederic Sowa 提交于 
      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>
      a26552af
  33. 14 8月, 2014 1 次提交
  35. 06 8月, 2014 2 次提交
    • W
      net-timestamp: ACK timestamp for bytestreams · e1c8a607
      由 Willem de Bruijn 提交于 
      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>
      e1c8a607
    • N
      tcp: reduce spurious retransmits due to transient SACK reneging · 5ae344c9
      由 Neal Cardwell 提交于 
      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>
      5ae344c9
  37. 16 7月, 2014 1 次提交
  39. 08 7月, 2014 1 次提交
    • Y
      tcp: fix false undo corner cases · 6e08d5e3
      由 Yuchung Cheng 提交于 
      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>
      6e08d5e3
  41. 30 6月, 2014 1 次提交
  43. 28 6月, 2014 1 次提交
  45. 20 6月, 2014 1 次提交
    • N
      tcp: fix tcp_match_skb_to_sack() for unaligned SACK at end of an skb · 2cd0d743
      由 Neal Cardwell 提交于 
      If there is an MSS change (or misbehaving receiver) that causes a SACK
to arrive that covers the end of an skb but is less than one MSS, then
tcp_match_skb_to_sack() was rounding up pkt_len to the full length of
the skb ("Round if necessary..."), then chopping all bytes off the skb
and creating a zero-byte skb in the write queue.

This was visible now because the recently simplified TLP logic in
bef1909e ("tcp: fixing TLP's FIN recovery") could find that 0-byte
skb at the end of the write queue, and now that we do not check that
skb's length we could send it as a TLP probe.

Consider the following example scenario:

 mss: 1000
 skb: seq: 0 end_seq: 4000  len: 4000
 SACK: start_seq: 3999 end_seq: 4000

The tcp_match_skb_to_sack() code will compute:

 in_sack = false
 pkt_len = start_seq - TCP_SKB_CB(skb)->seq = 3999 - 0 = 3999
 new_len = (pkt_len / mss) * mss = (3999/1000)*1000 = 3000
 new_len += mss = 4000

Previously we would find the new_len > skb->len check failing, so we
would fall through and set pkt_len = new_len = 4000 and chop off
pkt_len of 4000 from the 4000-byte skb, leaving a 0-byte segment
afterward in the write queue.

With this new commit, we notice that the new new_len >= skb->len check
succeeds, so that we return without trying to fragment.

Fixes: adb92db8 ("tcp: Make SACK code to split only at mss boundaries")
Reported-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Ilpo Jarvinen <ilpo.jarvinen@helsinki.fi>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      2cd0d743
  47. 11 6月, 2014 1 次提交
  49. 03 6月, 2014 1 次提交
  51. 04 5月, 2014 1 次提交
  53. 21 4月, 2014 1 次提交
  55. 12 4月, 2014 1 次提交
    • D
      net: Fix use after free by removing length arg from sk_data_ready callbacks. · 676d2369
      由 David S. Miller 提交于 
      Several spots in the kernel perform a sequence like:

	skb_queue_tail(&sk->s_receive_queue, skb);
	sk->sk_data_ready(sk, skb->len);

But at the moment we place the SKB onto the socket receive queue it
can be consumed and freed up.  So this skb->len access is potentially
to freed up memory.

Furthermore, the skb->len can be modified by the consumer so it is
possible that the value isn't accurate.

And finally, no actual implementation of this callback actually uses
the length argument.  And since nobody actually cared about it's
value, lots of call sites pass arbitrary values in such as '0' and
even '1'.

So just remove the length argument from the callback, that way there
is no confusion whatsoever and all of these use-after-free cases get
fixed as a side effect.

Based upon a patch by Eric Dumazet and his suggestion to audit this
issue tree-wide.
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      676d2369