IPv6 needs a cookie in dst_check() call.
We need to add rx_dst_cookie and provide a family independent
sk_rx_dst_set(sk, skb) method to properly support IPv6 TCP early demux.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Introduce sk_gfp_atomic(), this function allows to inject sock specific
flags to each sock related allocation. It is only used on allocation
paths that may be required for writing pages back to network storage.
[davem@davemloft.net: Use sk_gfp_atomic only when necessary]
Signed-off-by: Peter Zijlstra <a.p.zijlstra@chello.nl>
Signed-off-by: Mel Gorman <mgorman@suse.de>
Acked-by: David S. Miller <davem@davemloft.net>
Cc: Neil Brown <neilb@suse.de>
Cc: Mike Christie <michaelc@cs.wisc.edu>
Cc: Eric B Munson <emunson@mgebm.net>
Cc: Eric Dumazet <eric.dumazet@gmail.com>
Cc: Sebastian Andrzej Siewior <sebastian@breakpoint.cc>
Cc: Mel Gorman <mgorman@suse.de>
Cc: Christoph Lameter <cl@linux.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This is the IPv6 missing bits for infrastructure added in commit
41063e9dd1 (ipv4: Early TCP socket demux.)
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
ICMP messages generated in output path if frame length is bigger than
mtu are actually lost because socket is owned by user (doing the xmit)
One example is the ipgre_tunnel_xmit() calling
icmp_send(skb, ICMP_DEST_UNREACH, ICMP_FRAG_NEEDED, htonl(mtu));
We had a similar case fixed in commit a34a101e1e (ipv6: disable GSO on
sockets hitting dst_allfrag).
Problem of such fix is that it relied on retransmit timers, so short tcp
sessions paid a too big latency increase price.
This patch uses the tcp_release_cb() infrastructure so that MTU
reduction messages (ICMP messages) are not lost, and no extra delay
is added in TCP transmits.
Reported-by: Maciej Żenczykowski <maze@google.com>
Diagnosed-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Nandita Dukkipati <nanditad@google.com>
Cc: Tom Herbert <therbert@google.com>
Cc: Tore Anderson <tore@fud.no>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch impelements the common code for both the client and server.
1. TCP Fast Open option processing. Since Fast Open does not have an
option number assigned by IANA yet, it shares the experiment option
code 254 by implementing draft-ietf-tcpm-experimental-options
with a 16 bits magic number 0xF989. This enables global experiments
without clashing the scarce(2) experimental options available for TCP.
When the draft status becomes standard (maybe), the client should
switch to the new option number assigned while the server supports
both numbers for transistion.
2. The new sysctl tcp_fastopen
3. A place holder init function
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This will be used so that we can compose a full flow key.
Even though we have a route in this context, we need more. In the
future the routes will be without destination address, source address,
etc. keying. One ipv4 route will cover entire subnets, etc.
In this environment we have to have a way to possess persistent storage
for redirects and PMTU information. This persistent storage will exist
in the FIB tables, and that's why we'll need to be able to rebuild a
full lookup flow key here. Using that flow key will do a fib_lookup()
and create/update the persistent entry.
Signed-off-by: David 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: Eric 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: David S. Miller <davem@davemloft.net>
opt always equals np->opts, so it is meaningless to define opt, and
check if opt does not equal np->opts and then try to free opt.
Signed-off-by: RongQing.Li <roy.qing.li@gmail.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
The code in tcp_v6_conn_request() was implicitly assuming that
tcp_v6_send_synack() would take care of dst_release(), much as
tcp_v4_send_synack() already does. This resulted in
tcp_v6_conn_request() leaking a dst if sysctl_tw_recycle is enabled.
This commit restructures tcp_v6_send_synack() so that it accepts a dst
pointer and takes care of releasing the dst that is passed in, to plug
the leak and avoid future surprises by bringing the IPv6 behavior in
line with the IPv4 side.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
With the recent change (earlier in this patch series) to set
flowi6_oif to treq->iif in inet6_csk_route_req(), the dst lookup in
these two functions is now identical, so tcp_v6_send_synack() can now
just call inet6_csk_route_req(), to reduce code duplication and keep
things closer to the IPv4 side, which is structured this way.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This commit changes inet_csk_route_req() so that it uses a pointer to
a struct flowi6, rather than allocating its own on the stack. This
brings its behavior in line with its IPv4 cousin,
inet_csk_route_req(), and allows a follow-on patch to fix a dst leak.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
drivers/net/caif/caif_hsi.c
drivers/net/usb/qmi_wwan.c
The qmi_wwan merge was trivial.
The caif_hsi.c, on the other hand, was not. It's a conflict between
1c385f1fdf ("caif-hsi: Replace platform
device with ops structure.") in the net-next tree and commit
39abbaef19 ("caif-hsi: Postpone init of
HIS until open()") in the net tree.
I did my best with that one and will ask Sjur to check it out.
Signed-off-by: David S. Miller <davem@davemloft.net>
If security_inet_conn_request() returns non-zero then TCP/IPv6 should
drop the request, just as in TCP/IPv4 and DCCP in both IPv4 and IPv6.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
One tricky issue on the ipv6 side vs. ipv4 is that the ICMP callouts
to handle the error pass the 32-bit info cookie in network byte order
whereas ipv4 passes it around in host byte order.
Like the ipv4 side, we have two helper functions. One for when we
have a socket context and one for when we do not.
ip6ip6 tunnels are not handled here, because they handle PMTU events
by essentially relaying another ICMP packet-too-big message back to
the original sender.
This patch allows us to get rid of rt6_do_pmtu_disc(). It handles all
kinds of situations that simply cannot happen when we do the PMTU
update directly using a fully resolved route.
In fact, the "plen == 128" check in ip6_rt_update_pmtu() can very
likely be removed or changed into a BUG_ON() check. We should never
have a prefixed ipv6 route when we get there.
Another piece of strange history here is that TCP and DCCP, unlike in
ipv4, never invoke the update_pmtu() method from their ICMP error
handlers. This is incredibly astonishing since this is the context
where we have the most accurate context in which to make a PMTU
update, namely we have a fully connected socket and associated cached
socket route.
Signed-off-by: David S. Miller <davem@davemloft.net>
Since it's guarenteed that we will access the inetpeer if we're trying
to do timewait recycling and TCP options were enabled on the
connection, just cache the peer in the timewait socket.
In the future, inetpeer lookups will be context dependent (per routing
realm), and this helps facilitate that as well.
Signed-off-by: David S. Miller <davem@davemloft.net>
The get_peer method TCP uses is full of special cases that make no
sense accommodating, and it also gets in the way of doing more
reasonable things here.
First of all, if the socket doesn't have a usable cached route, there
is no sense in trying to optimize timewait recycling.
Likewise for the case where we have IP options, such as SRR enabled,
that make the IP header destination address (and thus the destination
address of the route key) differ from that of the connection's
destination address.
Just return a NULL peer in these cases, and thus we're also able to
get rid of the clumsy inetpeer release logic.
Signed-off-by: David S. Miller <davem@davemloft.net>
There's a lot of places that open-code rt{,6}_get_peer() only because
they want to set 'create' to one. So add an rt{,6}_get_peer_create()
for their sake.
There were also a few spots open-coding plain rt{,6}_get_peer() and
those are transformed here as well.
Signed-off-by: David S. Miller <davem@davemloft.net>
add struct net as a parameter of inet_getpeer_v[4,6],
use net to replace &init_net.
and modify some places to provide net for inet_getpeer_v[4,6]
Signed-off-by: Gao feng <gaofeng@cn.fujitsu.com>
Signed-off-by: David 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: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
While testing how linux behaves on SYNFLOOD attack on multiqueue device
(ixgbe), I found that SYNACK messages were dropped at Qdisc level
because we send them all on a single queue.
Obvious choice is to reflect incoming SYN packet @queue_mapping to
SYNACK packet.
Under stress, my machine could only send 25.000 SYNACK per second (for
200.000 incoming SYN per second). NIC : ixgbe with 16 rx/tx queues.
After patch, not a single SYNACK is dropped.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Hans Schillstrom <hans.schillstrom@ericsson.com>
Cc: Jesper Dangaard Brouer <brouer@redhat.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Tom Herbert <therbert@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
bool conversions where possible.
__inline__ -> inline
space cleanups
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David 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: Joe Perches <joe@perches.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
It appears some networks play bad games with the two bits reserved for
ECN. This can trigger false congestion notifications and very slow
transferts.
Since RFC 3168 (6.1.1) forbids SYN packets to carry CT bits, we can
disable TCP ECN negociation if it happens we receive mangled CT bits in
the SYN packet.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Perry Lorier <perryl@google.com>
Cc: Matt Mathis <mattmathis@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Wilmer van der Gaast <wilmer@google.com>
Cc: Ankur Jain <jankur@google.com>
Cc: Tom Herbert <therbert@google.com>
Cc: Dave Täht <dave.taht@bufferbloat.net>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David 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: Tore Anderson <tore@fud.no>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Maciej Żenczykowski <maze@google.com>
Cc: Tom Herbert <therbert@google.com>
Tested-by: Tore Anderson <tore@fud.no>
Signed-off-by: David S. Miller <davem@davemloft.net>
Fix merge between commit 3adadc08cc ("net ax25: Reorder ax25_exit to
remove races") and commit 0ca7a4c87d ("net ax25: Simplify and
cleanup the ax25 sysctl handling")
The former moved around the sysctl register/unregister calls, the
later simply removed them.
With help from Stephen Rothwell.
Signed-off-by: David S. Miller <davem@davemloft.net>
While investigating TCP performance problems on 10Gb+ links, we found a
tcp sender was dropping lot of incoming ACKS because of sk_rcvbuf limit
in sk_add_backlog(), especially if receiver doesnt use GRO/LRO and sends
one ACK every two MSS segments.
A sender usually tweaks sk_sndbuf, but sk_rcvbuf stays at its default
value (87380), allowing a too small backlog.
A TCP ACK, even being small, can consume nearly same truesize space than
outgoing packets. Using sk_rcvbuf + sk_sndbuf as a limit makes sense and
is fast to compute.
Performance results on netperf, single flow, receiver with disabled
GRO/LRO : 7500 Mbits instead of 6050 Mbits, no more TCPBacklogDrop
increments at sender.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Tom Herbert <therbert@google.com>
Cc: Maciej Żenczykowski <maze@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Cc: Rick Jones <rick.jones2@hp.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
sk_add_backlog() & sk_rcvqueues_full() hard coded sk_rcvbuf as the
memory limit. We need to make this limit a parameter for TCP use.
No functional change expected in this patch, all callers still using the
old sk_rcvbuf limit.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Tom Herbert <therbert@google.com>
Cc: Maciej Żenczykowski <maze@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Cc: Rick Jones <rick.jones2@hp.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
net/ipv4/tcp_ipv4.c: In function 'tcp_v4_init_sock':
net/ipv4/tcp_ipv4.c:1891:19: warning: unused variable 'tp' [-Wunused-variable]
net/ipv6/tcp_ipv6.c: In function 'tcp_v6_init_sock':
net/ipv6/tcp_ipv6.c:1836:19: warning: unused variable 'tp' [-Wunused-variable]
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: David S. Miller <davem@davemloft.net>
Commit f5fff5d forgot to fix TCP_MAXSEG behavior IPv6 sockets, so IPv6
TCP server sockets that used TCP_MAXSEG would find that the advmss of
child sockets would be incorrect. This commit mirrors the advmss logic
from tcp_v4_syn_recv_sock in tcp_v6_syn_recv_sock. Eventually this
logic should probably be shared between IPv4 and IPv6, but this at
least fixes this issue.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This commit moves the (substantial) common code shared between
tcp_v4_init_sock() and tcp_v6_init_sock() to a new address-family
independent function, tcp_init_sock().
Centralizing this functionality should help avoid drift issues,
e.g. where the IPv4 side is updated without a corresponding update to
IPv6. There was already some drift: IPv4 initialized snd_cwnd to
TCP_INIT_CWND, while the IPv6 side was still initializing snd_cwnd to
2 (in this case it should not matter, since snd_cwnd is also
initialized in tcp_init_metrics(), but the general risks and
maintenance overhead remain).
When diffing the old and new code, note that new tcp_init_sock()
function uses the order of steps from the tcp_v4_init_sock()
implementation (the order is slightly different in
tcp_v6_init_sock()).
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
When we need to clone skb, we dont drop a packet.
Call consume_skb() to not confuse dropwatch.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
1/ regression fix for Xen as it now trips over a broken assumption
about the dma address size on 32-bit builds
2/ new quirk for netdma to ignore dma channels that cannot meet
netdma alignment requirements
3/ fixes for two long standing issues in ioatdma (ring size overflow)
and iop-adma (potential stack corruption)
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=F3IR
-----END PGP SIGNATURE-----
Merge tag 'dmaengine-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/djbw/dmaengine
Pull dmaengine fixes from Dan Williams:
1/ regression fix for Xen as it now trips over a broken assumption
about the dma address size on 32-bit builds
2/ new quirk for netdma to ignore dma channels that cannot meet
netdma alignment requirements
3/ fixes for two long standing issues in ioatdma (ring size overflow)
and iop-adma (potential stack corruption)
* tag 'dmaengine-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/djbw/dmaengine:
netdma: adding alignment check for NETDMA ops
ioatdma: DMA copy alignment needed to address IOAT DMA silicon errata
ioat: ring size variables need to be 32bit to avoid overflow
iop-adma: Corrected array overflow in RAID6 Xscale(R) test.
ioat: fix size of 'completion' for Xen
This is the fallout from adding memcpy alignment workaround for certain
IOATDMA hardware. NetDMA will only use DMA engine that can handle byte align
ops.
Acked-by: David S. Miller <davem@davemloft.net>
Signed-off-by: Dave Jiang <dave.jiang@intel.com>
Signed-off-by: Dan Williams <dan.j.williams@intel.com>
Currently, it is not easily possible to get TOS/DSCP value of packets from
an incoming TCP stream. The mechanism is there, IP_PKTOPTIONS getsockopt
with IP_RECVTOS set, the same way as incoming TTL can be queried. This is
not actually implemented for TOS, though.
This patch adds this functionality, both for IPv4 (IP_PKTOPTIONS) and IPv6
(IPV6_2292PKTOPTIONS). For IPv4, like in the IP_RECVTTL case, the value of
the TOS field is stored from the other party's ACK.
This is needed for proxies which require DSCP transparency. One such example
is at http://zph.bratcheda.org/.
Signed-off-by: Jiri Benc <jbenc@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
TCP RST mechanism is broken in TCP md5(RFC2385). When
connection is gone, md5 key is lost, sending RST
without md5 hash is deem to ignored by peer. This can
be a problem since RST help protocal like bgp to fast
recove from peer crash.
In most case, users of tcp md5, such as bgp and ldp,
have listener on both sides to accept connection from peer.
md5 keys for peers are saved in listening socket.
There are two cases in finding md5 key when connection is
lost:
1.Passive receive RST: The message is send to well known port,
tcp will associate it with listner. md5 key is gotten from
listener.
2.Active receive RST (no sock): The message is send to ative
side, there is no socket associated with the message. In this
case, finding listener from source port, then find md5 key from
listener.
we are not loosing sercuriy here:
packet is checked with md5 hash. No RST is generated
if md5 hash doesn't match or no md5 key can be found.
Signed-off-by: Shawn Lu <shawn.lu@ericsson.com>
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch makes sure we use appropriate memory barriers before
publishing tp->md5sig_info, allowing tcp_md5_do_lookup() being used from
tcp_v4_send_reset() without holding socket lock (upcoming patch from
Shawn Lu)
Note we also need to respect rcu grace period before its freeing, since
we can free socket without this grace period thanks to
SLAB_DESTROY_BY_RCU
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Cc: Shawn Lu <shawn.lu@ericsson.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
In order to be able to support proper RST messages for TCP MD5 flows, we
need to allow access to MD5 keys without locking listener socket.
This conversion is a nice cleanup, and shrinks size of timewait sockets
by 80 bytes.
IPv6 code reuses generic code found in IPv4 instead of duplicating it.
Control path uses GFP_KERNEL allocations instead of GFP_ATOMIC.
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Cc: Shawn Lu <shawn.lu@ericsson.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
We no longer use md5_add() method from struct tcp_sock_af_ops
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
md5 key is added in socket through remote address.
remote address should be used in finding md5 key when
sending out reset packet.
Signed-off-by: shawnlu <shawn.lu@ericsson.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch allows each namespace to independently set up
its levels for tcp memory pressure thresholds. This patch
alone does not buy much: we need to make this values
per group of process somehow. This is achieved in the
patches that follows in this patchset.
Signed-off-by: Glauber Costa <glommer@parallels.com>
Reviewed-by: KAMEZAWA Hiroyuki <kamezawa.hiroyu@jp.fujitsu.com>
CC: David S. Miller <davem@davemloft.net>
CC: Eric W. Biederman <ebiederm@xmission.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch introduces memory pressure controls for the tcp
protocol. It uses the generic socket memory pressure code
introduced in earlier patches, and fills in the
necessary data in cg_proto struct.
Signed-off-by: Glauber Costa <glommer@parallels.com>
Reviewed-by: KAMEZAWA Hiroyuki <kamezawa.hiroyu@jp.fujtisu.com>
CC: Eric W. Biederman <ebiederm@xmission.com>
Signed-off-by: David 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: Glauber Costa <glommer@parallels.com>
Reviewed-by: Hiroyouki 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: David S. Miller <davem@davemloft.net>