Socket state LAST_ACK should allow TSQ to send additional frames,
or else we rely on incoming ACKS or timers to send them.
Reported-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Matt Mathis <mattmathis@google.com>
Cc: Mahesh Bandewar <maheshb@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This introduce TSQ (TCP Small Queues)

TSQ goal is to reduce number of TCP packets in xmit queues (qdisc &
device queues), to reduce RTT and cwnd bias, part of the bufferbloat
problem.

sk->sk_wmem_alloc not allowed to grow above a given limit,
allowing no more than ~128KB [1] per tcp socket in qdisc/dev layers at a
given time.

TSO packets are sized/capped to half the limit, so that we have two
TSO packets in flight, allowing better bandwidth use.

As a side effect, setting the limit to 40000 automatically reduces the
standard gso max limit (65536) to 40000/2 : It can help to reduce
latencies of high prio packets, having smaller TSO packets.

This means we divert sock_wfree() to a tcp_wfree() handler, to
queue/send following frames when skb_orphan() [2] is called for the
already queued skbs.

Results on my dev machines (tg3/ixgbe nics) are really impressive,
using standard pfifo_fast, and with or without TSO/GSO.

Without reduction of nominal bandwidth, we have reduction of buffering
per bulk sender :
< 1ms on Gbit (instead of 50ms with TSO)
< 8ms on 100Mbit (instead of 132 ms)

I no longer have 4 MBytes backlogged in qdisc by a single netperf
session, and both side socket autotuning no longer use 4 Mbytes.

As skb destructor cannot restart xmit itself ( as qdisc lock might be
taken at this point ), we delegate the work to a tasklet. We use one
tasklest per cpu for performance reasons.

If tasklet finds a socket owned by the user, it sets TSQ_OWNED flag.
This flag is tested in a new protocol method called from release_sock(),
to eventually send new segments.

[1] New /proc/sys/net/ipv4/tcp_limit_output_bytes tunable
[2] skb_orphan() is usually called at TX completion time,
  but some drivers call it in their start_xmit() handler.
  These drivers should at least use BQL, or else a single TCP
  session can still fill the whole NIC TX ring, since TSQ will
  have no effect.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Dave Taht <dave.taht@bufferbloat.net>
Cc: Tom Herbert <therbert@google.com>
Cc: Matt Mathis <mattmathis@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

tcp_make_synack() clones the dst, and callers release it.

We can avoid two atomic operations per SYNACK if tcp_make_synack()
consumes dst instead of cloning it.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

There is no value using sock_wmalloc() in tcp_make_synack().

A listener socket only sends SYNACK packets, they are not queued in a
socket queue, only in Qdisc and device layers, so the number of in
flight packets is limited in these layers. We used sock_wmalloc() with
the %force parameter set to 1 to ignore socket limits anyway.

This patch removes two atomic operations per SYNACK packet.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

bool conversions where possible.

__inline__ -> inline

space cleanups
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Use the current debugging style and enable dynamic_debug.
Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Standardize the net core ratelimited logging functions.

Coalesce formats, align arguments.
Change a printk then vprintk sequence to use printf extension %pV.
Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Implementing the advanced early retransmit (sysctl_tcp_early_retrans==2).
Delays the fast retransmit by an interval of RTT/4. We borrow the
RTO timer to implement the delay. If we receive another ACK or send
a new packet, the timer is cancelled and restored to original RTO
value offset by time elapsed.  When the delayed-ER timer fires,
we enter fast recovery and perform fast retransmit.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Quoting Tore Anderson from :
https://bugzilla.kernel.org/show_bug.cgi?id=42572

When RTAX_FEATURE_ALLFRAG is set on a route, the effective TCP segment
size does not take into account the size of the IPv6 Fragmentation
header that needs to be included in outbound packets, causing every
transmitted TCP segment to be fragmented across two IPv6 packets, the
latter of which will only contain 8 bytes of actual payload.

RTAX_FEATURE_ALLFRAG is typically set on a route in response to
receving a ICMPv6 Packet Too Big message indicating a Path MTU of less
than 1280 bytes. 1280 bytes is the minimum IPv6 MTU, however ICMPv6
PTBs with MTU < 1280 are still valid, in particular when an IPv6
packet is sent to an IPv4 destination through a stateless translator.
Any ICMPv4 Need To Fragment packets originated from the IPv4 part of
the path will be translated to ICMPv6 PTB which may then indicate an
MTU of less than 1280.

The Linux kernel refuses to reduce the effective MTU to anything below
1280 bytes, instead it sets it to exactly 1280 bytes, and
RTAX_FEATURE_ALLFRAG is also set. However, the TCP segment size appears
to be set to 1240 bytes (1280 Path MTU - 40 bytes of IPv6 header),
instead of 1232 (additionally taking into account the 8 bytes required
by the IPv6 Fragmentation extension header).

This in turn results in rather inefficient transmission, as every
transmitted TCP segment now is split in two fragments containing
1232+8 bytes of payload.

After this patch, all the outgoing packets that includes a
Fragmentation header all are "atomic" or "non-fragmented" fragments,
i.e., they both have Offset=0 and More Fragments=0.

With help from David S. Miller
Reported-by: NTore Anderson <tore@fud.no>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Maciej Żenczykowski <maze@google.com>
Cc: Tom Herbert <therbert@google.com>
Tested-by: NTore Anderson <tore@fud.no>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Reading queues under repair mode is done with recvmsg call.
The queue-under-repair set by TCP_REPAIR_QUEUE option is used
to determine which queue should be read. Thus both send and
receive queue can be read with this.

Caller must pass the MSG_PEEK flag.

Writing to queues is done with sendmsg call and yet again --
the repair-queue option can be used to push data into the
receive queue.

When putting an skb into receive queue a zero tcp header is
appented to its head to address the tcp_hdr(skb)->syn and
the ->fin checks by the (after repair) tcp_recvmsg. These
flags flags are both set to zero and that's why.

The fin cannot be met in the queue while reading the source
socket, since the repair only works for closed/established
sockets and queueing fin packet always changes its state.

The syn in the queue denotes that the respective skb's seq
is "off-by-one" as compared to the actual payload lenght. Thus,
at the rcv queue refill we can just drop this flag and set the
skb's sequences to precice values.

When the repair mode is turned off, the write queue seqs are
updated so that the whole queue is considered to be 'already sent,
waiting for ACKs' (write_seq = snd_nxt <= snd_una). From the
protocol POV the send queue looks like it was sent, but the data
between the write_seq and snd_nxt is lost in the network.

This helps to avoid another sockoption for setting the snd_nxt
sequence. Leaving the whole queue in a 'not yet sent' state (as
it will be after sendmsg-s) will not allow to receive any acks
from the peer since the ack_seq will be after the snd_nxt. Thus
even the ack for the window probe will be dropped and the
connection will be 'locked' with the zero peer window.
Signed-off-by: NPavel Emelyanov <xemul@parallels.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This includes (according the the previous description):

* TCP_REPAIR sockoption

This one just puts the socket in/out of the repair mode.
Allowed for CAP_NET_ADMIN and for closed/establised sockets only.
When repair mode is turned off and the socket happens to be in
the established state the window probe is sent to the peer to
'unlock' the connection.

* TCP_REPAIR_QUEUE sockoption

This one sets the queue which we're about to repair. The
'no-queue' is set by default.

* TCP_QUEUE_SEQ socoption

Sets the write_seq/rcv_nxt of a selected repaired queue.
Allowed for TCP_CLOSE-d sockets only. When the socket changes
its state the other seq-s are changed by the kernel according
to the protocol rules (most of the existing code is actually
reused).

* Ability to forcibly bind a socket to a port

The sk->sk_reuse is set to SK_FORCE_REUSE.

* Immediate connect modification

The connect syscall initializes the connection, then directly jumps
to the code which finalizes it.

* Silent close modification

The close just aborts the connection (similar to SO_LINGER with 0
time) but without sending any FIN/RST-s to peer.
Signed-off-by: NPavel Emelyanov <xemul@parallels.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This is just the preparation patch, which makes the needed for
TCP repair code ready for use.
Signed-off-by: NPavel Emelyanov <xemul@parallels.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Alexander Beregalov reported skb_over_panic errors and provided stack
trace.

I occurs commit a21d4572 (tcp: avoid order-1 allocations on wifi and
tx path) added a regression, when a retransmit is done after a partial
ACK.

tcp_retransmit_skb() tries to aggregate several frames if the first one
has enough available room to hold the following ones payload. This is
controlled by /proc/sys/net/ipv4/tcp_retrans_collapse tunable (default :
enabled)

Problem is we must make sure _pskb_trim_head() doesnt fool
skb_availroom() when pulling some bytes from skb (this pull is done when
receiver ACK part of the frame).
Reported-by: NAlexander Beregalov <a.beregalov@gmail.com>
Cc: Marc MERLIN <marc@merlins.org>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Use of "unsigned int" is preferred to bare "unsigned" in net tree.
Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Marc Merlin reported many order-1 allocations failures in TX path on its
wireless setup, that dont make any sense with MTU=1500 network, and non
SG capable hardware.

After investigation, it turns out TCP uses sk_stream_alloc_skb() and
used as a convention skb_tailroom(skb) to know how many bytes of data
payload could be put in this skb (for non SG capable devices)

Note : these skb used kmalloc-4096 (MTU=1500 + MAX_HEADER +
sizeof(struct skb_shared_info) being above 2048)

Later, mac80211 layer need to add some bytes at the tail of skb
(IEEE80211_ENCRYPT_TAILROOM = 18 bytes) and since no more tailroom is
available has to call pskb_expand_head() and request order-1
allocations.

This patch changes sk_stream_alloc_skb() so that only
sk->sk_prot->max_header bytes of headroom are reserved, and use a new
skb field, avail_size to hold the data payload limit.

This way, order-0 allocations done by TCP stack can leave more than 2 KB
of tailroom and no more allocation is performed in mac80211 layer (or
any layer needing some tailroom)

avail_size is unioned with mark/dropcount, since mark will be set later
in IP stack for output packets. Therefore, skb size is unchanged.
Reported-by: NMarc MERLIN <marc@merlins.org>
Tested-by: NMarc MERLIN <marc@merlins.org>
Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This commit fixes tcp_trim_head() to recalculate the number of
segments in the skb with the skb's existing MSS, so trimming the head
causes the skb segment count to be monotonically non-increasing - it
should stay the same or go down, but not increase.

Previously tcp_trim_head() used the current MSS of the connection. But
if there was a decrease in MSS between original transmission and ACK
(e.g. due to PMTUD), this could cause tcp_trim_head() to
counter-intuitively increase the segment count when trimming bytes off
the head of an skb. This violated assumptions in tcp_tso_acked() that
tcp_trim_head() only decreases the packet count, so that packets_acked
in tcp_tso_acked() could underflow, leading tcp_clean_rtx_queue() to
pass u32 pkts_acked values as large as 0xffffffff to
ca_ops->pkts_acked().

As an aside, if tcp_trim_head() had really wanted the skb to reflect
the current MSS, it should have called tcp_set_skb_tso_segs()
unconditionally, since a decrease in MSS would mean that a
single-packet skb should now be sliced into multiple segments.
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NNandita Dukkipati <nanditad@google.com>
Acked-by: NIlpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

It might be useful to get a counter of failed tcp_retransmit_skb()
calls.
Reported-by: NSatoru Moriya <satoru.moriya@hds.com>
Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch replaces all uses of struct sock fields' memory_pressure,
memory_allocated, sockets_allocated, and sysctl_mem to acessor
macros. Those macros can either receive a socket argument, or a mem_cgroup
argument, depending on the context they live in.

Since we're only doing a macro wrapping here, no performance impact at all is
expected in the case where we don't have cgroups disabled.
Signed-off-by: NGlauber Costa <glommer@parallels.com>
Reviewed-by: NHiroyouki Kamezawa <kamezawa.hiroyu@jp.fujitsu.com>
CC: David S. Miller <davem@davemloft.net>
CC: Eric W. Biederman <ebiederm@xmission.com>
CC: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

commit f07d960d (tcp: avoid frag allocation for small frames)
breaked assumption in tcp stack that skb is either linear (skb->data_len
== 0), or fully fragged (skb->data_len == skb->len)

tcp_trim_head() made this assumption, we must fix it.

Thanks to Vijay for providing a very detailed explanation.
Reported-by: NVijay Subramanian <subramanian.vijay@gmail.com>
Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

We discovered that TCP stack could retransmit misaligned skbs if a
malicious peer acknowledged sub MSS frame. This currently can happen
only if output interface is non SG enabled : If SG is enabled, tcp
builds headless skbs (all payload is included in fragments), so the tcp
trimming process only removes parts of skb fragments, header stay
aligned.

Some arches cant handle misalignments, so force a head reallocation and
shrink headroom to MAX_TCP_HEADER.

Dont care about misaligments on x86 and PPC (or other arches setting
NET_IP_ALIGN to 0)

This patch introduces __pskb_copy() which can specify the headroom of
new head, and pskb_copy() becomes a wrapper on top of __pskb_copy()
Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Since 2005 (c1b4a7e6)
tcp_tso_should_defer has been using tcp_max_burst() as a target limit
for deciding how large to make outgoing TSO packets when not using
sysctl_tcp_tso_win_divisor. But since 2008
(dd9e0dda) tcp_max_burst() returns the
reordering degree. We should not have tcp_tso_should_defer attempt to
build larger segments just because there is more reordering. This
commit splits the notion of deferral size used in TSO from the notion
of burst size used in cwnd moderation, and returns the TSO deferral
limit to its original value.
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

TCP_NODELAY is weaker than TCP_CORK, when TCP_CORK was set, small
segments will always pass Nagle test regardless of TCP_NODELAY option.
Signed-off-by: NFeng King <kinwin2008@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Adding const qualifiers to pointers can ease code review, and spot some
bugs. It might allow compiler to optimize code further.

For example, is it legal to temporary write a null cksum into tcphdr
in tcp_md5_hash_header() ? I am afraid a sniffer could catch the
temporary null value...
Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

To ease skb->truesize sanitization, its better to be able to localize
all references to skb frags size.

Define accessors : skb_frag_size() to fetch frag size, and
skb_frag_size_{set|add|sub}() to manipulate it.
Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Rename struct tcp_skb_cb "flags" to "tcp_flags" to ease code review and
maintenance.

Its content is a combination of FIN/SYN/RST/PSH/ACK/URG/ECE/CWR flags
Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch implements Proportional Rate Reduction (PRR) for TCP.
PRR is an algorithm that determines TCP's sending rate in fast
recovery. PRR avoids excessive window reductions and aims for
the actual congestion window size at the end of recovery to be as
close as possible to the window determined by the congestion control
algorithm. PRR also improves accuracy of the amount of data sent
during loss recovery.

The patch implements the recommended flavor of PRR called PRR-SSRB
(Proportional rate reduction with slow start reduction bound) and
replaces the existing rate halving algorithm. PRR improves upon the
existing Linux fast recovery under a number of conditions including:
  1) burst losses where the losses implicitly reduce the amount of
outstanding data (pipe) below the ssthresh value selected by the
congestion control algorithm and,
  2) losses near the end of short flows where application runs out of
data to send.

As an example, with the existing rate halving implementation a single
loss event can cause a connection carrying short Web transactions to
go into the slow start mode after the recovery. This is because during
recovery Linux pulls the congestion window down to packets_in_flight+1
on every ACK. A short Web response often runs out of new data to send
and its pipe reduces to zero by the end of recovery when all its packets
are drained from the network. Subsequent HTTP responses using the same
connection will have to slow start to raise cwnd to ssthresh. PRR on
the other hand aims for the cwnd to be as close as possible to ssthresh
by the end of recovery.

A description of PRR and a discussion of its performance can be found at
the following links:
- IETF Draft:
    http://tools.ietf.org/html/draft-mathis-tcpm-proportional-rate-reduction-01
- IETF Slides:
    http://www.ietf.org/proceedings/80/slides/tcpm-6.pdf
    http://tools.ietf.org/agenda/81/slides/tcpm-2.pdf
- Paper to appear in Internet Measurements Conference (IMC) 2011:
    Improving TCP Loss Recovery
    Nandita Dukkipati, Matt Mathis, Yuchung Cheng
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Signed-off-by: NIan Campbell <ian.campbell@citrix.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: "Pekka Savola (ipv6)" <pekkas@netcore.fi>
Cc: James Morris <jmorris@namei.org>
Cc: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
Cc: Patrick McHardy <kaber@trash.net>
Cc: netdev@vger.kernel.org
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This allows us to acquire the exact route keying information from the
protocol, however that might be managed.

It handles all of the possibilities, from the simplest case of storing
the key in inet->cork.fl to the more complex setup SCTP has where
individual transports determine the flow.
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

All callers are prepared for alloc failures anyway, so this error
can safely be boomeranged to the callers domain without super
bad consequences. ...At worst the connection might go into a state
where each RTO tries to (unsuccessfully) re-fragment with such
a mis-sized value and eventually dies.
Signed-off-by: NIlpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Fixes generated by 'codespell' and manually reviewed.
Signed-off-by: NLucas De Marchi <lucas.demarchi@profusion.mobi>

Fix a bug that undo_retrans is incorrectly decremented when undo_marker is
not set or undo_retrans is already 0. This happens when sender receives
more DSACK ACKs than packets retransmitted during the current
undo phase. This may also happen when sender receives DSACK after
the undo operation is completed or cancelled.

Fix another bug that undo_retrans is incorrectly incremented when
sender retransmits an skb and tcp_skb_pcount(skb) > 1 (TSO). This case
is rare but not impossible.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NIlpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch changes the default initial receive window to 10 mss
(defined constant). The default window is limited to the maximum
of 10*1460 and 2*mss (when mss > 1460).

draft-ietf-tcpm-initcwnd-00 is a proposal to the IETF that recommends
increasing TCP's initial congestion window to 10 mss or about 15KB.
Leading up to this proposal were several large-scale live Internet
experiments with an initial congestion window of 10 mss (IW10), where
we showed that the average latency of HTTP responses improved by
approximately 10%. This was accompanied by a slight increase in
retransmission rate (0.5%), most of which is coming from applications
opening multiple simultaneous connections. To understand the extreme
worst case scenarios, and fairness issues (IW10 versus IW3), we further
conducted controlled testbed experiments. We came away finding minimal
negative impact even under low link bandwidths (dial-ups) and small
buffers.  These results are extremely encouraging to adopting IW10.

However, an initial congestion window of 10 mss is useless unless a TCP
receiver advertises an initial receive window of at least 10 mss.
Fortunately, in the large-scale Internet experiments we found that most
widely used operating systems advertised large initial receive windows
of 64KB, allowing us to experiment with a wide range of initial
congestion windows. Linux systems were among the few exceptions that
advertised a small receive window of 6KB. The purpose of this patch is
to fix this shortcoming.

References:
1. A comprehensive list of all IW10 references to date.
http://code.google.com/speed/protocols/tcpm-IW10.html

2. Paper describing results from large-scale Internet experiments with IW10.
http://ccr.sigcomm.org/drupal/?q=node/621

3. Controlled testbed experiments under worst case scenarios and a
fairness study.
http://www.ietf.org/proceedings/79/slides/tcpm-0.pdf

4. Raw test data from testbed experiments (Linux senders/receivers)
with initial congestion and receive windows of both 10 mss.
http://research.csc.ncsu.edu/netsrv/?q=content/iw10

5. Internet-Draft. Increasing TCP's Initial Window.
https://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Acked-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Make all RTAX_ADVMSS metric accesses go through a new helper function,
dst_metric_advmss().

Leave the actual default metric as "zero" in the real metric slot,
and compute the actual default value dynamically via a new dst_ops
AF specific callback.

For stacked IPSEC routes, we use the advmss of the path which
preserves existing behavior.

Unlike ipv4/ipv6, DecNET ties the advmss to the mtu and thus updates
advmss on pmtu updates.  This inconsistency in advmss handling
results in more raw metric accesses than I wish we ended up with.
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Make sure sysctl_tcp_cookie_size is read once in
tcp_cookie_size_check(), or we might return an illegal value to caller
if sysctl_tcp_cookie_size is changed by another cpu.
Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
Cc: Ben Hutchings <bhutchings@solarflare.com>
Cc: William Allen Simpson <william.allen.simpson@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

sysctl_tcp_tso_win_divisor might be set to zero while one cpu runs in
tcp_tso_should_defer(). Make sure we dont allow a divide by zero by
reading sysctl_tcp_tso_win_divisor exactly once.
Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The bug has to do with boundary checks on the initial receive window.
If the initial receive window falls between init_cwnd and the
receive window specified by the user, the initial window is incorrectly
brought down to init_cwnd. The correct behavior is to allow it to
remain unchanged.
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

TCP_BASE_MSS is defined, but not used.
commit 5d424d5a introduce this macro, so use
it to initial sysctl_tcp_base_mss.

commit 5d424d5a
Author: John Heffner <jheffner@psc.edu>
Date:   Mon Mar 20 17:53:41 2006 -0800

    [TCP]: MTU probing
Signed-off-by: NShan Wei <shanwei@cn.fujitsu.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

In dev_pick_tx, don't do work in calculating queue
index or setting
the index in the sock unless the device has more than one queue.  This
allows the sock to be set only with a queue index of a multi-queue
device which is desirable if device are stacked like in a tunnel.

We also allow the mapping of a socket to queue to be changed.  To
maintain in order packet transmission a flag (ooo_okay) has been
added to the sk_buff structure.  If a transport layer sets this flag
on a packet, the transmit queue can be changed for the socket.
Presumably, the transport would set this if there was no possbility
of creating OOO packets (for instance, there are no packets in flight
for the socket).  This patch includes the modification in TCP output
for setting this flag.
Signed-off-by: NTom Herbert <therbert@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The current tcp_connect code completely ignores errors from sending an skb.
This makes sense in many situations (like -ENOBUFFS) but I want to be able to
immediately fail connections if they are denied by the SELinux netfilter hook.
Netfilter does not normally return ECONNREFUSED when it drops a packet so we
respect that error code as a final and fatal error that can not be recovered.
Based-on-patch-by: NPatrick McHardy <kaber@trash.net>
Signed-off-by: NEric Paris <eparis@redhat.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

"gadget", "through", "command", "maintain", "maintain", "controller", "address",
"between", "initiali[zs]e", "instead", "function", "select", "already",
"equal", "access", "management", "hierarchy", "registration", "interest",
"relative", "memory", "offset", "already",
Signed-off-by: NUwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: NJiri Kosina <jkosina@suse.cz>