tcp_time_stamp will no longer be tied to jiffies.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Use tcp_jiffies32 instead of tcp_time_stamp, since
tcp_time_stamp will soon be only used for TCP TS option.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Use tcp_jiffies32 instead of tcp_time_stamp, since
tcp_time_stamp will soon be only used for TCP TS option.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Use tcp_jiffies32 instead of tcp_time_stamp to feed
tp->snd_cwnd_stamp.
tcp_time_stamp will soon be a litle bit more expensive
than simply reading 'jiffies'.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Use tcp_jiffies32 instead of tcp_time_stamp to feed
tp->lsndtime.
tcp_time_stamp will soon be a litle bit more expensive
than simply reading 'jiffies'.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Idea is to later convert tp->tcp_mstamp to a full u64 counter
using usec resolution, so that we can later have fine
grained TCP TS clock (RFC 7323), regardless of HZ value.
We try to refresh tp->tcp_mstamp only when necessary.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
BBR congestion control depends on pacing, and pacing is
currently handled by sch_fq packet scheduler for performance reasons,
and also because implemening pacing with FQ was convenient to truly
avoid bursts.
However there are many cases where this packet scheduler constraint
is not practical.
- Many linux hosts are not focusing on handling thousands of TCP
flows in the most efficient way.
- Some routers use fq_codel or other AQM, but still would like
to use BBR for the few TCP flows they initiate/terminate.
This patch implements an automatic fallback to internal pacing.
Pacing is requested either by BBR or use of SO_MAX_PACING_RATE option.
If sch_fq happens to be in the egress path, pacing is delegated to
the qdisc, otherwise pacing is done by TCP itself.
One advantage of pacing from TCP stack is to get more precise rtt
estimations, and less work done from TX completion, since TCP Small
queue limits are not generally hit. Setups with single TX queue but
many cpus might even benefit from this.
Note that unlike sch_fq, we do not take into account header sizes.
Taking care of these headers would add additional complexity for
no practical differences in behavior.
Some performance numbers using 800 TCP_STREAM flows rate limited to
~48 Mbit per second on 40Gbit NIC.
If MQ+pfifo_fast is used on the NIC :
$ sar -n DEV 1 5 | grep eth
14:48:44 eth0 725743.00 2932134.00 46776.76 4335184.68 0.00 0.00 1.00
14:48:45 eth0 725349.00 2932112.00 46751.86 4335158.90 0.00 0.00 0.00
14:48:46 eth0 725101.00 2931153.00 46735.07 4333748.63 0.00 0.00 0.00
14:48:47 eth0 725099.00 2931161.00 46735.11 4333760.44 0.00 0.00 1.00
14:48:48 eth0 725160.00 2931731.00 46738.88 4334606.07 0.00 0.00 0.00
Average: eth0 725290.40 2931658.20 46747.54 4334491.74 0.00 0.00 0.40
$ vmstat 1 5
procs -----------memory---------- ---swap-- -----io---- -system-- ------cpu-----
r b swpd free buff cache si so bi bo in cs us sy id wa st
4 0 0 259825920 45644 2708324 0 0 21 2 247 98 0 0 100 0 0
4 0 0 259823744 45644 2708356 0 0 0 0 2400825 159843 0 19 81 0 0
0 0 0 259824208 45644 2708072 0 0 0 0 2407351 159929 0 19 81 0 0
1 0 0 259824592 45644 2708128 0 0 0 0 2405183 160386 0 19 80 0 0
1 0 0 259824272 45644 2707868 0 0 0 32 2396361 158037 0 19 81 0 0
Now use MQ+FQ :
lpaa23:~# echo fq >/proc/sys/net/core/default_qdisc
lpaa23:~# tc qdisc replace dev eth0 root mq
$ sar -n DEV 1 5 | grep eth
14:49:57 eth0 678614.00 2727930.00 43739.13 4033279.14 0.00 0.00 0.00
14:49:58 eth0 677620.00 2723971.00 43674.69 4027429.62 0.00 0.00 1.00
14:49:59 eth0 676396.00 2719050.00 43596.83 4020125.02 0.00 0.00 0.00
14:50:00 eth0 675197.00 2714173.00 43518.62 4012938.90 0.00 0.00 1.00
14:50:01 eth0 676388.00 2719063.00 43595.47 4020171.64 0.00 0.00 0.00
Average: eth0 676843.00 2720837.40 43624.95 4022788.86 0.00 0.00 0.40
$ vmstat 1 5
procs -----------memory---------- ---swap-- -----io---- -system-- ------cpu-----
r b swpd free buff cache si so bi bo in cs us sy id wa st
2 0 0 259832240 46008 2710912 0 0 21 2 223 192 0 1 99 0 0
1 0 0 259832896 46008 2710744 0 0 0 0 1702206 198078 0 17 82 0 0
0 0 0 259830272 46008 2710596 0 0 0 0 1696340 197756 1 17 83 0 0
4 0 0 259829168 46024 2710584 0 0 16 0 1688472 197158 1 17 82 0 0
3 0 0 259830224 46024 2710408 0 0 0 0 1692450 197212 0 18 82 0 0
As expected, number of interrupts per second is very different.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Van Jacobson <vanj@google.com>
Cc: Jerry Chu <hkchu@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Congestion control modules that want full control over congestion
control behavior do not want the cwnd modifications controlled by
the sysctl_tcp_slow_start_after_idle code path.
So skip those code paths for CC modules that use the cong_control()
API.
As an example, those cwnd effects are not desired for the BBR congestion
control algorithm.
Fixes: c0402760f5 ("tcp: new CC hook to set sending rate with rate_sample in any CA state")
Signed-off-by: Wei Wang <weiwan@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Pull networking updates from David Millar:
"Here are some highlights from the 2065 networking commits that
happened this development cycle:
1) XDP support for IXGBE (John Fastabend) and thunderx (Sunil Kowuri)
2) Add a generic XDP driver, so that anyone can test XDP even if they
lack a networking device whose driver has explicit XDP support
(me).
3) Sparc64 now has an eBPF JIT too (me)
4) Add a BPF program testing framework via BPF_PROG_TEST_RUN (Alexei
Starovoitov)
5) Make netfitler network namespace teardown less expensive (Florian
Westphal)
6) Add symmetric hashing support to nft_hash (Laura Garcia Liebana)
7) Implement NAPI and GRO in netvsc driver (Stephen Hemminger)
8) Support TC flower offload statistics in mlxsw (Arkadi Sharshevsky)
9) Multiqueue support in stmmac driver (Joao Pinto)
10) Remove TCP timewait recycling, it never really could possibly work
well in the real world and timestamp randomization really zaps any
hint of usability this feature had (Soheil Hassas Yeganeh)
11) Support level3 vs level4 ECMP route hashing in ipv4 (Nikolay
Aleksandrov)
12) Add socket busy poll support to epoll (Sridhar Samudrala)
13) Netlink extended ACK support (Johannes Berg, Pablo Neira Ayuso,
and several others)
14) IPSEC hw offload infrastructure (Steffen Klassert)"
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/net-next: (2065 commits)
tipc: refactor function tipc_sk_recv_stream()
tipc: refactor function tipc_sk_recvmsg()
net: thunderx: Optimize page recycling for XDP
net: thunderx: Support for XDP header adjustment
net: thunderx: Add support for XDP_TX
net: thunderx: Add support for XDP_DROP
net: thunderx: Add basic XDP support
net: thunderx: Cleanup receive buffer allocation
net: thunderx: Optimize CQE_TX handling
net: thunderx: Optimize RBDR descriptor handling
net: thunderx: Support for page recycling
ipx: call ipxitf_put() in ioctl error path
net: sched: add helpers to handle extended actions
qed*: Fix issues in the ptp filter config implementation.
qede: Fix concurrency issue in PTP Tx path processing.
stmmac: Add support for SIMATIC IOT2000 platform
net: hns: fix ethtool_get_strings overflow in hns driver
tcp: fix wraparound issue in tcp_lp
bpf, arm64: fix jit branch offset related to ldimm64
bpf, arm64: implement jiting of BPF_XADD
...
Andrey found a way to trigger the WARN_ON_ONCE(delta < len) in
skb_try_coalesce() using syzkaller and a filter attached to a TCP
socket over loopback interface.
I believe one issue with looped skbs is that tcp_trim_head() can end up
producing skb with under estimated truesize.
It hardly matters for normal conditions, since packets sent over
loopback are never truncated.
Bytes trimmed from skb->head should not change skb truesize, since
skb->head is not reallocated.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts were simply overlapping changes. In the net/ipv4/route.c
case the code had simply moved around a little bit and the same fix
was made in both 'net' and 'net-next'.
In the net/sched/sch_generic.c case a fix in 'net' happened at
the same time that a new argument was added to qdisc_hash_add().
Signed-off-by: David S. Miller <davem@davemloft.net>
Because TCP_MIB_OUTRSTS is an important count, so always increase it
whatever send it successfully or not.
Now move the increment of TCP_MIB_OUTRSTS to the top of
tcp_send_active_reset to make sure it is increased always even though
fail to alloc skb.
Signed-off-by: Gao Feng <fgao@ikuai8.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Define one new macro TCP_MAX_WSCALE instead of literal number '14',
and use U16_MAX instead of 65535 as the max value of TCP window.
There is another minor change, use rounddown(space, mss) instead of
(space / mss) * mss;
Signed-off-by: Gao Feng <fgao@ikuai8.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
1. Move the "window = tp->rcv_wnd;" into the condition block without
tp->rx_opt.rcv_wscale.
Because it is unnecessary when enable wscale;
2. Use the macro ALIGN instead of two statements.
The two statements are used to make window align to 1<<wscale.
Use the ALIGN is more clearer.
3. Use the rounddown to make codes clearer.
Signed-off-by: Gao Feng <fgao@ikuai8.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Prevent sending out a left-shifted sequence number from a Linux sender in
response to a peer's shrunk receive-window caused by losing least significant
bits in window-scaling.
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: James Morris <jmorris@namei.org>
Cc: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
Cc: Patrick McHardy <kaber@trash.net>
Signed-off-by: Cheng Cui <Cheng.Cui@netapp.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
When same struct dst_entry can be used for many different
neighbours we can not use it for pending confirmations.
Use the new sk_dst_confirm() helper to propagate the
indication from received packets to sock_confirm_neigh().
Reported-by: YueHaibing <yuehaibing@huawei.com>
Fixes: 5110effee8 ("net: Do delayed neigh confirmation.")
Fixes: f2bb4bedf3 ("ipv4: Cache output routes in fib_info nexthops.")
Tested-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Julian Anastasov <ja@ssi.bg>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Josef Bacik diagnosed following problem :
I was seeing random disconnects while testing NBD over loopback.
This turned out to be because NBD sets pfmemalloc on it's socket,
however the receiving side is a user space application so does not
have pfmemalloc set on its socket. This means that
sk_filter_trim_cap will simply drop this packet, under the
assumption that the other side will simply retransmit. Well we do
retransmit, and then the packet is just dropped again for the same
reason.
It seems the better way to address this problem is to clear pfmemalloc
in the TCP transmit path. pfmemalloc strict control really makes sense
on the receive path.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Josef Bacik <jbacik@fb.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Small cleanup factorizing code doing the TCP_MAXSEG clamping.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
syszkaller fuzzer was able to trigger a divide by zero, when
TCP window scaling is not enabled.
SO_RCVBUF can be used not only to increase sk_rcvbuf, also
to decrease it below current receive buffers utilization.
If mss is negative or 0, just return a zero TCP window.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reported-by: Dmitry Vyukov <dvyukov@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Currently the retransmission stats are not incremented if the
retransmit fails locally. But we always increment the other packet
counters that track total packet/bytes sent. Awkwardly while we
don't count these failed retransmits in RETRANSSEGS, we do count
them in FAILEDRETRANS.
If the qdisc is dropping many packets this could under-estimate
TCP retransmission rate substantially from both SNMP or per-socket
TCP_INFO stats. This patch changes this by always incrementing
retransmission stats on retransmission attempts and failures.
Another motivation is to properly track retransmists in
SCM_TIMESTAMPING_OPT_STATS. Since SCM_TSTAMP_SCHED collection is
triggered in tcp_transmit_skb(), If tp->total_retrans is incremented
after the function, we'll always mis-count by the amount of the
latest retransmission.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Refactor the cookie check logic in tcp_send_syn_data() into a function.
This function will be called else where in later changes.
Signed-off-by: Wei Wang <weiwan@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch removes the support of RFC5827 early retransmit (i.e.,
fast recovery on small inflight with <3 dupacks) because it is
subsumed by the new RACK loss detection. More specifically when
RACK receives DUPACKs, it'll arm a reordering timer to start fast
recovery after a quarter of (min)RTT, hence it covers the early
retransmit except RACK does not limit itself to specific inflight
or dupack numbers.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Forward retransmit is an esoteric feature in RFC3517 (condition(3)
in the NextSeg()). Basically if a packet is not considered lost by
the current criteria (# of dupacks etc), but the congestion window
has room for more packets, then retransmit this packet.
However it actually conflicts with the rest of recovery design. For
example, when reordering is detected we want to be conservative
in retransmitting packets but forward-retransmit feature would
break that to force more retransmission. Also the implementation is
fairly complicated inside the retransmission logic inducing extra
iterations in the write queue. With RACK losses are being detected
timely and this heuristic is no longer necessary. There this patch
removes the feature.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
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 patch makes RACK install a reordering timer when it suspects
some packets might be lost, but wants to delay the decision
a little bit to accomodate reordering.
It does not create a new timer but instead repurposes the existing
RTO timer, because both are meant to retransmit packets.
Specifically it arms a timer ICSK_TIME_REO_TIMEOUT when
the RACK timing check fails. The wait time is set to
RACK.RTT + RACK.reo_wnd - (NOW - Packet.xmit_time) + fudge
This translates to expecting a packet (Packet) should take
(RACK.RTT + RACK.reo_wnd + fudge) to deliver after it was sent.
When there are multiple packets that need a timer, we use one timer
with the maximum timeout. Therefore the timer conservatively uses
the maximum window to expire N packets by one timeout, instead of
N timeouts to expire N packets sent at different times.
The fudge factor is 2 jiffies to ensure when the timer fires, all
the suspected packets would exceed the deadline and be marked lost
by tcp_rack_detect_loss(). It has to be at least 1 jiffy because the
clock may tick between calling icsk_reset_xmit_timer(timeout) and
actually hang the timer. The next jiffy is to lower-bound the timeout
to 2 jiffies when reo_wnd is < 1ms.
When the reordering timer fires (tcp_rack_reo_timeout): If we aren't
in Recovery we'll enter fast recovery and force fast retransmit.
This is very similar to the early retransmit (RFC5827) except RACK
is not constrained to only enter recovery for small outstanding
flights.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
ktime is a union because the initial implementation stored the time in
scalar nanoseconds on 64 bit machine and in a endianess optimized timespec
variant for 32bit machines. The Y2038 cleanup removed the timespec variant
and switched everything to scalar nanoseconds. The union remained, but
become completely pointless.
Get rid of the union and just keep ktime_t as simple typedef of type s64.
The conversion was done with coccinelle and some manual mopping up.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Cc: Peter Zijlstra <peterz@infradead.org>
Madalin reported crashes happening in tcp_tasklet_func() on powerpc64
Before TSQ_QUEUED bit is cleared, we must ensure the changes done
by list_del(&tp->tsq_node); are committed to memory, otherwise
corruption might happen, as an other cpu could catch TSQ_QUEUED
clearance too soon.
We can notice that old kernels were immune to this bug, because
TSQ_QUEUED was cleared after a bh_lock_sock(sk)/bh_unlock_sock(sk)
section, but they could have missed a kick to write additional bytes,
when NIC interrupts for a given flow are spread to multiple cpus.
Affected TCP flows would need an incoming ACK or RTO timer to add more
packets to the pipe. So overall situation should be better now.
Fixes: b223feb9de ("tcp: tsq: add shortcut in tcp_tasklet_func()")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reported-by: Madalin Bucur <madalin.bucur@nxp.com>
Tested-by: Madalin Bucur <madalin.bucur@nxp.com>
Tested-by: Xing Lei <xing.lei@nxp.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
tsq_flags being in the same cache line than sk_wmem_alloc
makes a lot of sense. Both fields are changed from tcp_wfree()
and more generally by various TSQ related functions.
Prior patch made room in struct sock and added sk_tsq_flags,
this patch deletes tsq_flags from struct tcp_sock.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Adding a likely() in tcp_mtu_probe() moves its code which used to
be inlined in front of tcp_write_xmit()
We still have a cache line miss to access icsk->icsk_mtup.enabled,
we will probably have to reorganize fields to help data locality.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Always allow the two first skbs in write queue to be sent,
regardless of sk_wmem_alloc/sk_pacing_rate values.
This helps a lot in situations where TX completions are delayed either
because of driver latencies or softirq latencies.
Test is done with no cache line misses.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Under high load, tcp_wfree() has an atomic operation trying
to schedule a tasklet over and over.
We can schedule it only if our per cpu list was empty.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Under high stress, I've seen tcp_tasklet_func() consuming
~700 usec, handling ~150 tcp sockets.
By setting TCP_TSQ_DEFERRED in tcp_wfree(), we give a chance
for other cpus/threads entering tcp_write_xmit() to grab it,
allowing tcp_tasklet_func() to skip sockets that already did
an xmit cycle.
In the future, we might give to ACK processing an increased
budget to reduce even more tcp_tasklet_func() amount of work.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Instead of atomically clear TSQ_THROTTLED and atomically set TSQ_QUEUED
bits, use one cmpxchg() to perform a single locked operation.
Since the following patch will also set TCP_TSQ_DEFERRED here,
this cmpxchg() will make this addition free.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This is a cleanup, to ease code review of following patches.
Old 'enum tsq_flags' is renamed, and a new enumeration is added
with the flags used in cmpxchg() operations as opposed to
single bit operations.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
jiffies based timestamps allow for easy inference of number of devices
behind NAT translators and also makes tracking of hosts simpler.
commit ceaa1fef65 ("tcp: adding a per-socket timestamp offset")
added the main infrastructure that is needed for per-connection ts
randomization, in particular writing/reading the on-wire tcp header
format takes the offset into account so rest of stack can use normal
tcp_time_stamp (jiffies).
So only two items are left:
- add a tsoffset for request sockets
- extend the tcp isn generator to also return another 32bit number
in addition to the ISN.
Re-use of ISN generator also means timestamps are still monotonically
increasing for same connection quadruple, i.e. PAWS will still work.
Includes fixes from Eric Dumazet.
Signed-off-by: Florian Westphal <fw@strlen.de>
Acked-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch measures the amount of time when TCP runs out of new data
to send to the network due to insufficient send buffer, while TCP
is still busy delivering (i.e. write queue is not empty). The goal
is to indicate either the send buffer autotuning or user SO_SNDBUF
setting has resulted network under-utilization.
The measurement starts conservatively by checking various conditions
to minimize false claims (i.e. under-estimation is more likely).
The measurement stops when the SOCK_NOSPACE flag is cleared. But it
does not account the time elapsed till the next application write.
Also the measurement only starts if the sender is still busy sending
data, s.t. the limit accounted is part of the total busy time.
Signed-off-by: Francis Yan <francisyyan@gmail.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch measures the total time when the TCP stops sending because
the receiver's advertised window is not large enough. Note that
once the limit is lifted we are likely in the busy status if we
have data pending.
Signed-off-by: Francis Yan <francisyyan@gmail.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch measures TCP busy time, which is defined as the period
of time when sender has data (or FIN) to send. The time starts when
data is buffered and stops when the write queue is flushed by ACKs
or error events.
Note the busy time does not include SYN time, unless data is
included in SYN (i.e. Fast Open). It does include FIN time even
if the FIN carries no payload. Excluding pure FIN is possible but
would incur one additional test in the fast path, which may not
be worth it.
Signed-off-by: Francis Yan <francisyyan@gmail.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch implements the skeleton of the TCP chronograph
instrumentation on sender side limits:
1) idle (unspec)
2) busy sending data other than 3-4 below
3) rwnd-limited
4) sndbuf-limited
The limits are enumerated 'tcp_chrono'. Since a connection in
theory can idle forever, we do not track the actual length of this
uninteresting idle period. For the rest we track how long the sender
spends in each limit. At any point during the life time of a
connection, the sender must be in one of the four states.
If there are multiple conditions worthy of tracking in a chronograph
then the highest priority enum takes precedence over
the other conditions. So that if something "more interesting"
starts happening, stop the previous chrono and start a new one.
The time unit is jiffy(u32) in order to save space in tcp_sock.
This implies application must sample the stats no longer than every
49 days of 1ms jiffy.
Signed-off-by: Francis Yan <francisyyan@gmail.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
With TCP MTU probing enabled and offload TX checksumming disabled,
tcp_mtu_probe() calculated the wrong checksum when a fragment being copied
into the probe's SKB had an odd length. This was caused by the direct use
of skb_copy_and_csum_bits() to calculate the checksum, as it pads the
fragment being copied, if needed. When this fragment was not the last, a
subsequent call used the previous checksum without considering this
padding.
The effect was a stale connection in one way, as even retransmissions
wouldn't solve the problem, because the checksum was never recalculated for
the full SKB length.
Signed-off-by: Douglas Caetano dos Santos <douglascs@taghos.com.br>
Signed-off-by: David S. Miller <davem@davemloft.net>
Since the TFO socket is accepted right off SYN-data, the socket
owner can call getsockopt(TCP_INFO) to collect ongoing SYN-ACK
retransmission or timeout stats (i.e., tcpi_total_retrans,
tcpi_retransmits). Currently those stats are only updated
upon handshake completes. This patch fixes it.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch fixes these under-accounting SNMP rtx stats
LINUX_MIB_TCPFORWARDRETRANS
LINUX_MIB_TCPFASTRETRANS
LINUX_MIB_TCPSLOWSTARTRETRANS
when retransmitting TSO packets
Fixes: 10d3be5692 ("tcp-tso: do not split TSO packets at retransmit time")
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
We saw sch_fq drops caused by the per flow limit of 100 packets and TCP
when dealing with large cwnd and bursts of retransmits.
Even after increasing the limit to 1000, and even after commit
10d3be5692 ("tcp-tso: do not split TSO packets at retransmit time"),
we can still have these drops.
Under certain conditions, TCP can spend a considerable amount of
time queuing thousands of skbs in a single tcp_xmit_retransmit_queue()
invocation, incurring latency spikes and stalls of other softirq
handlers.
This patch implements TSQ for retransmits, limiting number of packets
and giving more chance for scheduling packets in both ways.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Export tcp_mss_to_mtu(), so that congestion control modules can use
this to help calculate a pacing rate.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
To allow congestion control modules to use the default TSO auto-sizing
algorithm as one of the ingredients in their own decision about TSO sizing:
1) Export tcp_tso_autosize() so that CC modules can use it.
2) Change tcp_tso_autosize() to allow callers to specify a minimum
number of segments per TSO skb, in case the congestion control
module has a different notion of the best floor for TSO skbs for
the connection right now. For very low-rate paths or policed
connections it can be appropriate to use smaller TSO skbs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Add the tso_segs_goal() function in tcp_congestion_ops to allow the
congestion control module to specify the number of segments that
should be in a TSO skb sent by tcp_write_xmit() and
tcp_xmit_retransmit_queue(). The congestion control module can either
request a particular number of segments in TSO skb that we transmit,
or return 0 if it doesn't care.
This allows the upcoming BBR congestion control module to select small
TSO skb sizes if the module detects that the bottleneck bandwidth is
very low, or that the connection is policed to a low rate.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
If a TCP socket gets a large write queue, an overflow can happen
in a test in __tcp_retransmit_skb() preventing all retransmits.
The flow then stalls and resets after timeouts.
Tested:
sysctl -w net.core.wmem_max=1000000000
netperf -H dest -- -s 1000000000
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
While chasing tcp_xmit_retransmit_queue() kasan issue, I found
that we could avoid reading sacked field of skb that we wont send,
possibly removing one cache line miss.
Very minor change in slow path, but why not ? ;)
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_select_initial_window() intends to advertise a window
scaling for the maximum possible window size. To do so,
it considers the maximum of net.ipv4.tcp_rmem[2] and
net.core.rmem_max as the only possible upper-bounds.
However, users with CAP_NET_ADMIN can use SO_RCVBUFFORCE
to set the socket's receive buffer size to values
larger than net.ipv4.tcp_rmem[2] and net.core.rmem_max.
Thus, SO_RCVBUFFORCE is effectively ignored by
tcp_select_initial_window().
To fix this, consider the maximum of net.ipv4.tcp_rmem[2],
net.core.rmem_max and socket's initial buffer space.
Fixes: b0573dea1f ("[NET]: Introduce SO_{SND,RCV}BUFFORCE socket options")
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Suggested-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Several cases of overlapping changes, except the packet scheduler
conflicts which deal with the addition of the free list parameter
to qdisc_enqueue().
Signed-off-by: David S. Miller <davem@davemloft.net>
Arjun reported a bug in TCP stack and bisected it to a recent commit.
In case where we process SACK, we can coalesce multiple skbs
into fat ones (tcp_shift_skb_data()), to lower write queue
overhead, because we do not expect to retransmit these packets.
However, SACK reneging can happen, forcing the sender to retransmit
all these packets. If skb->len is above 64KB, we then send buggy
IP packets that could hang TSO engine on cxgb4.
Neal suggested to use tcp_tso_autosize() instead of tp->gso_segs
so that we cook packets of optimal size vs TCP/pacing.
Thanks to Arjun for reporting the bug and running the tests !
Fixes: 10d3be5692 ("tcp-tso: do not split TSO packets at retransmit time")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reported-by: Arjun V <arjun@chelsio.com>
Tested-by: Arjun V <arjun@chelsio.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Add in_flight (bytes in flight when packet was sent) field
to tx component of tcp_skb_cb and make it available to
congestion modules' pkts_acked() function through the
ack_sample function argument.
Signed-off-by: Lawrence Brakmo <brakmo@fb.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_hdr() is slightly more expensive than using skb->data in contexts
where we know they point to the same byte.
In receive path, tcp_v4_rcv() and tcp_v6_rcv() are in this situation,
as tcp header has not been pulled yet.
In output path, the same can be said when we just pushed the tcp header
in the skb, in tcp_transmit_skb() and tcp_make_synack()
Also factorize the two checks for tcb->tcp_flags & TCPHDR_SYN in
tcp_transmit_skb() and pass tcp header pointer to tcp_ecn_send(),
so that compiler can further optimize and avoid a reload.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
The nf_conntrack_core.c fix in 'net' is not relevant in 'net-next'
because we no longer have a per-netns conntrack hash.
The ip_gre.c conflict as well as the iwlwifi ones were cases of
overlapping changes.
Conflicts:
drivers/net/wireless/intel/iwlwifi/mvm/tx.c
net/ipv4/ip_gre.c
net/netfilter/nf_conntrack_core.c
Signed-off-by: David S. Miller <davem@davemloft.net>
In the very unlikely case __tcp_retransmit_skb() can not use the cloning
done in tcp_transmit_skb(), we need to refresh skb_mstamp before doing
the copy and transmit, otherwise TCP TS val will be an exact copy of
original transmit.
Fixes: 7faee5c0d5 ("tcp: remove TCP_SKB_CB(skb)->when")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Hosts sending lot of ACK packets exhibit high sock_wfree() cost
because of cache line miss to test SOCK_USE_WRITE_QUEUE
We could move this flag close to sk_wmem_alloc but it is better
to perform the atomic_sub_and_test() on a clean cache line,
as it avoid one extra bus transaction.
skb_orphan_partial() can also have a fast track for packets that either
are TCP acks, or already went through another skb_orphan_partial()
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
We want to to make TCP stack preemptible, as draining prequeue
and backlog queues can take lot of time.
Many SNMP updates were assuming that BH (and preemption) was disabled.
Need to convert some __NET_INC_STATS() calls to NET_INC_STATS()
and some __TCP_INC_STATS() to TCP_INC_STATS()
Before using this_cpu_ptr(net->ipv4.tcp_sk) in tcp_v4_send_reset()
and tcp_v4_send_ack(), we add an explicit preempt disabled section.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
The SKBTX_ACK_TSTAMP flag is set in skb_shinfo->tx_flags when
the timestamp of the TCP acknowledgement should be reported on
error queue. Since accessing skb_shinfo is likely to incur a
cache-line miss at the time of receiving the ack, the
txstamp_ack bit was added in tcp_skb_cb, which is set iff
the SKBTX_ACK_TSTAMP flag is set for an skb. This makes
SKBTX_ACK_TSTAMP flag redundant.
Remove the SKBTX_ACK_TSTAMP and instead use the txstamp_ack bit
everywhere.
Note that this frees one bit in shinfo->tx_flags.
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Martin KaFai Lau <kafai@fb.com>
Suggested-by: Willem de Bruijn <willemb@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Rename NET_INC_STATS_BH() to __NET_INC_STATS()
and NET_ADD_STATS_BH() to __NET_ADD_STATS()
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Linux TCP stack painfully segments all TSO/GSO packets before retransmits.
This was fine back in the days when TSO/GSO were emerging, with their
bugs, but we believe the dark age is over.
Keeping big packets in write queues, but also in stack traversal
has a lot of benefits.
- Less memory overhead, because write queues have less skbs
- Less cpu overhead at ACK processing.
- Better SACK processing, as lot of studies mentioned how
awful linux was at this ;)
- Less cpu overhead to send the rtx packets
(IP stack traversal, netfilter traversal, drivers...)
- Better latencies in presence of losses.
- Smaller spikes in fq like packet schedulers, as retransmits
are not constrained by TCP Small Queues.
1 % packet losses are common today, and at 100Gbit speeds, this
translates to ~80,000 losses per second.
Losses are often correlated, and we see many retransmit events
leading to 1-MSS train of packets, at the time hosts are already
under stress.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts were two cases of simple overlapping changes,
nothing serious.
In the UDP case, we need to add a hlist_add_tail_rcu()
to linux/rculist.h, because we've moved UDP socket handling
away from using nulls lists.
Signed-off-by: David S. Miller <davem@davemloft.net>
When removing sk_refcnt manipulation on synflood, I missed that
using skb_set_owner_w() was racy, if sk->sk_wmem_alloc had already
transitioned to 0.
We should hold sk_refcnt instead, but this is a big deal under attack.
(Doing so increase performance from 3.2 Mpps to 3.8 Mpps only)
In this patch, I chose to not attach a socket to syncookies skb.
Performance is now 5 Mpps instead of 3.2 Mpps.
Following patch will remove last known false sharing in
tcp_rcv_state_process()
Fixes: 3b24d854cb ("tcp/dccp: do not touch listener sk_refcnt under synflood")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Per RFC4898, they count segments sent/received
containing a positive length data segment (that includes
retransmission segments carrying data). Unlike
tcpi_segs_out/in, tcpi_data_segs_out/in excludes segments
carrying no data (e.g. pure ack).
The patch also updates the segs_in in tcp_fastopen_add_skb()
so that segs_in >= data_segs_in property is kept.
Together with retransmission data, tcpi_data_segs_out
gives a better signal on the rxmit rate.
v6: Rebase on the latest net-next
v5: Eric pointed out that checking skb->len is still needed in
tcp_fastopen_add_skb() because skb can carry a FIN without data.
Hence, instead of open coding segs_in and data_segs_in, tcp_segs_in()
helper is used. Comment is added to the fastopen case to explain why
segs_in has to be reset and tcp_segs_in() has to be called before
__skb_pull().
v4: Add comment to the changes in tcp_fastopen_add_skb()
and also add remark on this case in the commit message.
v3: Add const modifier to the skb parameter in tcp_segs_in()
v2: Rework based on recent fix by Eric:
commit a9d99ce28e ("tcp: fix tcpi_segs_in after connection establishment")
Signed-off-by: Martin KaFai Lau <kafai@fb.com>
Cc: Chris Rapier <rapier@psc.edu>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Marcelo Ricardo Leitner <mleitner@redhat.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
There won't be any separate counters for socket memory consumed by
protocols other than TCP in the future. Remove the indirection and link
sockets directly to their owning memory cgroup.
Signed-off-by: Johannes Weiner <hannes@cmpxchg.org>
Reviewed-by: Vladimir Davydov <vdavydov@virtuozzo.com>
Acked-by: David S. Miller <davem@davemloft.net>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
There won't be a tcp control soft limit, so integrating the memcg code
into the global skmem limiting scheme complicates things unnecessarily.
Replace this with simple and clear charge and uncharge calls--hidden
behind a jump label--to account skb memory.
Note that this is not purely aesthetic: as a result of shoehorning the
per-memcg code into the same memory accounting functions that handle the
global level, the old code would compare the per-memcg consumption
against the smaller of the per-memcg limit and the global limit. This
allowed the total consumption of multiple sockets to exceed the global
limit, as long as the individual sockets stayed within bounds. After
this change, the code will always compare the per-memcg consumption to
the per-memcg limit, and the global consumption to the global limit, and
thus close this loophole.
Without a soft limit, the per-memcg memory pressure state in sockets is
generally questionable. However, we did it until now, so we continue to
enter it when the hard limit is hit, and packets are dropped, to let
other sockets in the cgroup know that they shouldn't grow their transmit
windows, either. However, keep it simple in the new callback model and
leave memory pressure lazily when the next packet is accepted (as
opposed to doing it synchroneously when packets are processed). When
packets are dropped, network performance will already be in the toilet,
so that should be a reasonable trade-off.
As described above, consumption is now checked on the per-memcg level
and the global level separately. Likewise, memory pressure states are
maintained on both the per-memcg level and the global level, and a
socket is considered under pressure when either level asserts as much.
Signed-off-by: Johannes Weiner <hannes@cmpxchg.org>
Reviewed-by: Vladimir Davydov <vdavydov@virtuozzo.com>
Acked-by: David S. Miller <davem@davemloft.net>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Conflicts:
drivers/net/geneve.c
Here we had an overlapping change, where in 'net' the extraneous stats
bump was being removed whilst in 'net-next' the final argument to
udp_tunnel6_xmit_skb() was being changed.
Signed-off-by: David S. Miller <davem@davemloft.net>
Yuchung tracked a regression caused by commit 57be5bdad7 ("ip: convert
tcp_sendmsg() to iov_iter primitives") for TCP Fast Open.
Some Fast Open users do not actually add any data in the SYN packet.
Fixes: 57be5bdad7 ("ip: convert tcp_sendmsg() to iov_iter primitives")
Reported-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Al Viro <viro@zeniv.linux.org.uk>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
If tcp_send_ack() can not allocate skb, we properly handle this
and setup a timer to try later.
Use __GFP_NOWARN to avoid polluting syslog in the case host is
under memory pressure, so that pertinent messages are not lost under
a flood of useless information.
sk_gfp_atomic() can use its gfp_mask argument (all callers currently
were using GFP_ATOMIC before this patch)
We rename sk_gfp_atomic() to sk_gfp_mask() to clearly express this
function now takes into account its second argument (gfp_mask)
Note that when tcp_transmit_skb() is called with clone_it set to false,
we do not attempt memory allocations, so can pass a 0 gfp_mask, which
most compilers can emit faster than a non zero or constant value.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
skb_set_owner_w() is called from various places that assume
skb->sk always point to a full blown socket (as it changes
sk->sk_wmem_alloc)
We'd like to attach skb to request sockets, and in the future
to timewait sockets as well. For these kind of pseudo sockets,
we need to take a traditional refcount and use sock_edemux()
as the destructor.
It is now time to un-inline skb_set_owner_w(), being too big.
Fixes: ca6fb06518 ("tcp: attach SYNACK messages to request sockets instead of listener")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Bisected-by: Haiyang Zhang <haiyangz@microsoft.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
net/ipv6/xfrm6_output.c
net/openvswitch/flow_netlink.c
net/openvswitch/vport-gre.c
net/openvswitch/vport-vxlan.c
net/openvswitch/vport.c
net/openvswitch/vport.h
The openvswitch conflicts were overlapping changes. One was
the egress tunnel info fix in 'net' and the other was the
vport ->send() op simplification in 'net-next'.
The xfrm6_output.c conflicts was also a simplification
overlapping a bug fix.
Signed-off-by: David S. Miller <davem@davemloft.net>
Remove the existing lost retransmit detection because RACK subsumes
it completely. This also stops the overloading the ack_seq field of
the skb control block.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
At the time of commit fff3269907 ("tcp: reflect SYN queue_mapping into
SYNACK packets") we had little ways to cope with SYN floods.
We no longer need to reflect incoming skb queue mappings, and instead
can pick a TX queue based on cpu cooking the SYNACK, with normal XPS
affinities.
Note that all SYNACK retransmits were picking TX queue 0, this no longer
is a win given that SYNACK rtx are now distributed on all cpus.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
One 32bit hole is following skc_refcnt, use it.
skc_incoming_cpu can also be an union for request_sock rcv_wnd.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
If a listen backlog is very big (to avoid syncookies), then
the listener sk->sk_wmem_alloc is the main source of false
sharing, as we need to touch it twice per SYNACK re-transmit
and TX completion.
(One SYN packet takes listener lock once, but up to 6 SYNACK
are generated)
By attaching the skb to the request socket, we remove this
source of contention.
Tested:
listen(fd, 10485760); // single listener (no SO_REUSEPORT)
16 RX/TX queue NIC
Sustain a SYNFLOOD attack of ~320,000 SYN per second,
Sending ~1,400,000 SYNACK per second.
Perf profiles now show listener spinlock being next bottleneck.
20.29% [kernel] [k] queued_spin_lock_slowpath
10.06% [kernel] [k] __inet_lookup_established
5.12% [kernel] [k] reqsk_timer_handler
3.22% [kernel] [k] get_next_timer_interrupt
3.00% [kernel] [k] tcp_make_synack
2.77% [kernel] [k] ipt_do_table
2.70% [kernel] [k] run_timer_softirq
2.50% [kernel] [k] ip_finish_output
2.04% [kernel] [k] cascade
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Application limited streams such as thin streams, that transmit small
amounts of payload in relatively few packets per RTT, can be prevented
from growing the CWND when in congestion avoidance. This leads to
increased sojourn times for data segments in streams that often transmit
time-dependent data.
Currently, a connection is considered CWND limited only after having
successfully transmitted at least one packet with new data, while at the
same time failing to transmit some unsent data from the output queue
because the CWND is full. Applications that produce small amounts of
data may be left in a state where it is never considered to be CWND
limited, because all unsent data is successfully transmitted each time
an incoming ACK opens up for more data to be transmitted in the send
window.
Fix by always testing whether the CWND is fully used after successful
packet transmissions, such that a connection is considered CWND limited
whenever the CWND has been filled. This is the correct behavior as
specified in RFC2861 (section 3.1).
Cc: Andreas Petlund <apetlund@simula.no>
Cc: Carsten Griwodz <griff@simula.no>
Cc: Jonas Markussen <jonassm@ifi.uio.no>
Cc: Kenneth Klette Jonassen <kennetkl@ifi.uio.no>
Cc: Mads Johannessen <madsjoh@ifi.uio.no>
Signed-off-by: Bendik Rønning Opstad <bro.devel+kernel@gmail.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Tested-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Tested-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
net/ipv4/arp.c
The net/ipv4/arp.c conflict was one commit adding a new
local variable while another commit was deleting one.
Signed-off-by: David S. Miller <davem@davemloft.net>
This is done to make sure we do not change listener socket
while sending SYNACK packets while socket lock is not held.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
listener socket is not locked when tcp_make_synack() is called.
We better make sure no field is written.
There is one exception : Since SYNACK packets are attached to the listener
at this moment (or SYN_RECV child in case of Fast Open),
sock_wmalloc() needs to update sk->sk_wmem_alloc, but this is done using
atomic operations so this is safe.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
SYNACK packets might be sent without holding socket lock.
For DCTCP/ECN sake, we should call INET_ECN_xmit() while
socket lock is owned, and only when we init/change congestion control.
This also fixies a bug if congestion module is changed from
dctcp to another one on a listener : we now clear ECN bits
properly.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
RST packets sent on behalf of TCP connections with TS option (RFC 7323
TCP timestamps) have incorrect TS val (set to 0), but correct TS ecr.
A > B: Flags [S], seq 0, win 65535, options [mss 1000,nop,nop,TS val 100
ecr 0], length 0
B > A: Flags [S.], seq 2444755794, ack 1, win 28960, options [mss
1460,nop,nop,TS val 7264344 ecr 100], length 0
A > B: Flags [.], ack 1, win 65535, options [nop,nop,TS val 110 ecr
7264344], length 0
B > A: Flags [R.], seq 1, ack 1, win 28960, options [nop,nop,TS val 0
ecr 110], length 0
We need to call skb_mstamp_get() to get proper TS val,
derived from skb->skb_mstamp
Note that RFC 1323 was advocating to not send TS option in RST segment,
but RFC 7323 recommends the opposite :
Once TSopt has been successfully negotiated, that is both <SYN> and
<SYN,ACK> contain TSopt, the TSopt MUST be sent in every non-<RST>
segment for the duration of the connection, and SHOULD be sent in an
<RST> segment (see Section 5.2 for details)
Note this RFC recommends to send TS val = 0, but we believe it is
premature : We do not know if all TCP stacks are properly
handling the receive side :
When an <RST> segment is
received, it MUST NOT be subjected to the PAWS check by verifying an
acceptable value in SEG.TSval, and information from the Timestamps
option MUST NOT be used to update connection state information.
SEG.TSecr MAY be used to provide stricter <RST> acceptance checks.
In 5 years, if/when all TCP stack are RFC 7323 ready, we might consider
to decide to send TS val = 0, if it buys something.
Fixes: 7faee5c0d5 ("tcp: remove TCP_SKB_CB(skb)->when")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch makes TLP to use 1 sec timer by default when RTT is
not available due to SYN/ACK retransmission or SYN cookies.
Prior to this change, the lack of RTT prevents TLP so the first
data packets sent can only be recovered by fast recovery or RTO.
If the fast recovery fails to trigger the RTO is 3 second when
SYN/ACK is retransmitted. With this patch we can trigger fast
recovery in 1sec instead.
Note that we need to check Fast Open more properly. A Fast Open
connection could be (accepted then) closed before it receives
the final ACK of 3WHS so the state is FIN_WAIT_1. Without the
new check, TLP will retransmit FIN instead of SYN/ACK.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
In commit b73c3d0e4f ("net: Save TX flow hash in sock and set in skbuf
on xmit"), Tom provided a l4 hash to most outgoing TCP packets.
We'd like to provide one as well for SYNACK packets, so that all packets
of a given flow share same txhash, to later enable bonding driver to
also use skb->hash to perform slave selection.
Note that a SYNACK retransmit shuffles the tx hash, as Tom did
in commit 265f94ff54 ("net: Recompute sk_txhash on negative routing
advice") for established sockets.
This has nice effect making TCP flows resilient to some kind of black
holes, even at connection establish phase.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Tom Herbert <tom@herbertland.com>
Cc: Mahesh Bandewar <maheshb@google.com>
Acked-by: Tom Herbert <tom@herbertland.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Issuing a CC TX_START event on control frames like pure ACK
is a waste of time, as a CC should not care.
Following patch needs this change, as we want CUBIC to properly track
idle time at a low cost, with a single TX_START being generated.
Yuchung might slightly refine the condition triggering TX_START
on a followup patch.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Cc: Jana Iyengar <jri@google.com>
Cc: Stephen Hemminger <stephen@networkplumber.org>
Cc: Sangtae Ha <sangtae.ha@gmail.com>
Cc: Lawrence Brakmo <lawrence@brakmo.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
slow start after idle might reduce cwnd, but we perform this
after first packet was cooked and sent.
With TSO/GSO, it means that we might send a full TSO packet
even if cwnd should have been reduced to IW10.
Moving the SSAI check in skb_entail() makes sense, because
we slightly reduce number of times this check is done,
especially for large send() and TCP Small queue callbacks from
softirq context.
As Neal pointed out, we also need to perform the check
if/when receive window opens.
Tested:
Following packetdrill test demonstrates the problem
// Test of slow start after idle
`sysctl -q net.ipv4.tcp_slow_start_after_idle=1`
0.000 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 65535 <mss 1000,sackOK,nop,nop,nop,wscale 7>
+0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6>
+.100 < . 1:1(0) ack 1 win 511
+0 accept(3, ..., ...) = 4
+0 setsockopt(4, SOL_SOCKET, SO_SNDBUF, [200000], 4) = 0
+0 write(4, ..., 26000) = 26000
+0 > . 1:5001(5000) ack 1
+0 > . 5001:10001(5000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10 }%
+.100 < . 1:1(0) ack 10001 win 511
+0 %{ assert tcpi_snd_cwnd == 20, tcpi_snd_cwnd }%
+0 > . 10001:20001(10000) ack 1
+0 > P. 20001:26001(6000) ack 1
+.100 < . 1:1(0) ack 26001 win 511
+0 %{ assert tcpi_snd_cwnd == 36, tcpi_snd_cwnd }%
+4 write(4, ..., 20000) = 20000
// If slow start after idle works properly, we should send 5 MSS here (cwnd/2)
+0 > . 26001:31001(5000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }%
+0 > . 31001:36001(5000) ack 1
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
When TLP fails to send new packet because of receive window
limit, it should fall back to retransmit the last packet instead.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
If TLP was unable to send a probe, it extended the RTO to
now + icsk_rto. But extending the RTO makes little sense
if no TLP probe went out. With this commit, instead of
extending the RTO we re-arm it relative to the transmit time
of the write queue head.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
While doing experiments with reordering resilience, we found
linux senders were not able to send at full speed under reordering,
because every incoming SACK was releasing one MSS.
This patch removes the limitation, as we did for CWR state
in commit a0ea700e40 ("tcp: tso: allow CA_CWR state in
tcp_tso_should_defer()")
Neal Cardwell had a concern about limited transmit so
Yuchung conducted experiments on GFE and found nothing
worth adding an extra check on fast path :
if (icsk->icsk_ca_state == TCP_CA_Disorder &&
tcp_sk(sk)->reordering == sysctl_tcp_reordering)
goto send_now;
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
V1 of this patch contains Eric Dumazet's suggestion to move the per
dst RTAX_QUICKACK check into tcp_in_quickack_mode(). Thanks Eric.
I ran some tests and after setting the "ip route change quickack 1"
knob there were still many delayed ACKs sent. This occured
because when icsk_ack.quick=0 the !icsk_ack.pingpong value is
subsequently ignored as tcp_in_quickack_mode() checks both these
values. The condition for a quick ack to trigger requires
that both icsk_ack.quick != 0 and icsk_ack.pingpong=0. Currently
only icsk_ack.pingpong is controlled by the knob. But the
icsk_ack.quick value changes dynamically depending on heuristics.
The crux of the matter is that delayed acks still cannot be entirely
disabled even with the RTAX_QUICKACK per dst knob enabled. This
patch ensures that a quick ack is always sent when the RTAX_QUICKACK
per dst knob is turned on.
The "ip route change quickack 1" knob was recently added to enable
quickacks. It was modeled around the TCP_QUICKACK setsockopt() option.
This issue is that even with "ip route change quickack 1" enabled
we still see delayed ACKs under some conditions. It would be nice
to be able to completely disable delayed ACKs.
Here is an example:
# netstat -s|grep dela
3 delayed acks sent
For all routes enable the knob
# ip route change quickack 1
Generate some traffic across a slow link and we still see the delayed
acks.
# netstat -s|grep dela
106 delayed acks sent
1 delayed acks further delayed because of locked socket
The issue is that both the "ip route change quickack 1" knob and
the TCP_QUICKACK option set the icsk_ack.pingpong variable to 0.
However at the business end in the __tcp_ack_snd_check() routine,
tcp_in_quickack_mode() checks that both icsk_ack.quick != 0
and icsk_ack.pingpong=0 in order to trigger a quickack. As
icsk_ack.quick is determined by heuristics it can be 0. When
that occurs the icsk_ack.pingpong value is ignored and a delayed
ACK is sent regardless.
This patch moves the RTAX_QUICKACK per dst check into the
tcp_in_quickack_mode() routine which ensures that a quickack is
always sent when the quickack knob is enabled for that dst.
Signed-off-by: Jon Maxwell <jmaxwell37@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
We had various issues in the past when TCP stack was modifying
gso_size/gso_segs while clones were in flight.
Commit c52e2421f7 ("tcp: must unclone packets before mangling them")
fixed these bugs and added a WARN_ON_ONCE(skb_cloned(skb)); in
tcp_set_skb_tso_segs()
These bugs are now fixed, and because TCP stack now only sets
shinfo->gso_size|segs on the clone itself, the check can be removed.
As a result of this change, compiler inlines tcp_set_skb_tso_segs() in
tcp_init_tso_segs()
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
In commit cd7d8498c9 ("tcp: change tcp_skb_pcount() location") we stored
gso_segs in a temporary cache hot location.
This patch does the same for gso_size.
This allows to save 2 cache line misses in tcp xmit path for
the last packet that is considered but not sent because of
various conditions (cwnd, tso defer, receiver window, TSQ...)
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_set_skb_tso_segs() & tcp_init_tso_segs() no longer
use the sock pointer.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Our goal is to touch skb_shinfo(skb) only when absolutely needed,
to avoid two cache line misses in TCP output path for last skb
that is considered but not sent because of various conditions
(cwnd, tso defer, receiver window, TSQ...)
A packet is GSO only when skb_shinfo(skb)->gso_size is not zero.
We can set skb_shinfo(skb)->gso_type to sk->sk_gso_type even for
non GSO packets.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Xen virtual network driver has higher latency than a physical NIC.
Having only 128K as limit for TSQ introduced 30% regression in guest
throughput.
This patch raises the limit to 256K. This reduces the regression to 8%.
This buys us more time to work out a proper solution in the long run.
Signed-off-by: Wei Liu <wei.liu2@citrix.com>
Cc: David Miller <davem@davemloft.net>
Cc: Eric Dumazet <eric.dumazet@gmail.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
By making sure sk->sk_gso_max_segs minimal value is one,
and sysctl_tcp_min_tso_segs minimal value is one as well,
tcp_tso_autosize() will return a non zero value.
We can then revert 843925f33f
("tcp: Do not apply TSO segment limit to non-TSO packets")
and save few cpu cycles in fast path.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Herbert Xu <herbert@gondor.apana.org.au>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Herbert Xu <herbert@gondor.apana.org.au>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch tracks the total number of inbound and outbound segments on a
TCP socket. One may use this number to have an idea on connection
quality when compared against the retransmissions.
RFC4898 named these : tcpEStatsPerfSegsIn and tcpEStatsPerfSegsOut
These are a 32bit field each 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->segs_out was placed near tp->snd_nxt for good data
locality and minimal performance impact, while tp->segs_in was placed
near tp->bytes_received for the same reason.
Join work with Eric Dumazet.
Note that received SYN are accounted on the listener, but sent SYNACK
are not accounted.
Signed-off-by: Marcelo Ricardo Leitner <mleitner@redhat.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
In commit 8e4d980ac2 ("tcp: fix behavior for epoll edge trigger")
we fixed a possible hang of TCP sockets under memory pressure,
by allowing sk_stream_alloc_skb() to use sk_forced_mem_schedule()
if no packet is in socket write queue.
It turns out there are other cases where we want to force memory
schedule :
tcp_fragment() & tso_fragment() need to split a big TSO packet into
two smaller ones. If we block here because of TCP memory pressure,
we can effectively block TCP socket from sending new data.
If no further ACK is coming, this hang would be definitive, and socket
has no chance to effectively reduce its memory usage.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This work as a follow-up of commit f7b3bec6f5 ("net: allow setting ecn
via routing table") and adds RFC3168 section 6.1.1.1. fallback for outgoing
ECN connections. In other words, this work adds a retry with a non-ECN
setup SYN packet, as suggested from the RFC on the first timeout:
[...] A host that receives no reply to an ECN-setup SYN within the
normal SYN retransmission timeout interval MAY resend the SYN and
any subsequent SYN retransmissions with CWR and ECE cleared. [...]
Schematic client-side view when assuming the server is in tcp_ecn=2 mode,
that is, Linux default since 2009 via commit 255cac91c3 ("tcp: extend
ECN sysctl to allow server-side only ECN"):
1) Normal ECN-capable path:
SYN ECE CWR ----->
<----- SYN ACK ECE
ACK ----->
2) Path with broken middlebox, when client has fallback:
SYN ECE CWR ----X crappy middlebox drops packet
(timeout, rtx)
SYN ----->
<----- SYN ACK
ACK ----->
In case we would not have the fallback implemented, the middlebox drop
point would basically end up as:
SYN ECE CWR ----X crappy middlebox drops packet
(timeout, rtx)
SYN ECE CWR ----X crappy middlebox drops packet
(timeout, rtx)
SYN ECE CWR ----X crappy middlebox drops packet
(timeout, rtx)
In any case, it's rather a smaller percentage of sites where there would
occur such additional setup latency: it was found in end of 2014 that ~56%
of IPv4 and 65% of IPv6 servers of Alexa 1 million list would negotiate
ECN (aka tcp_ecn=2 default), 0.42% of these webservers will fail to connect
when trying to negotiate with ECN (tcp_ecn=1) due to timeouts, which the
fallback would mitigate with a slight latency trade-off. Recent related
paper on this topic:
Brian Trammell, Mirja Kühlewind, Damiano Boppart, Iain Learmonth,
Gorry Fairhurst, and Richard Scheffenegger:
"Enabling Internet-Wide Deployment of Explicit Congestion Notification."
Proc. PAM 2015, New York.
http://ecn.ethz.ch/ecn-pam15.pdf
Thus, when net.ipv4.tcp_ecn=1 is being set, the patch will perform RFC3168,
section 6.1.1.1. fallback on timeout. For users explicitly not wanting this
which can be in DC use case, we add a net.ipv4.tcp_ecn_fallback knob that
allows for disabling the fallback.
tp->ecn_flags are not being cleared in tcp_ecn_clear_syn() on output, but
rather we let tcp_ecn_rcv_synack() take that over on input path in case a
SYN ACK ECE was delayed. Thus a spurious SYN retransmission will not prevent
ECN being negotiated eventually in that case.
Reference: https://www.ietf.org/proceedings/92/slides/slides-92-iccrg-1.pdf
Reference: https://www.ietf.org/proceedings/89/slides/slides-89-tsvarea-1.pdf
Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Mirja Kühlewind <mirja.kuehlewind@tik.ee.ethz.ch>
Signed-off-by: Brian Trammell <trammell@tik.ee.ethz.ch>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Dave That <dave.taht@gmail.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Introduce an optimized version of sk_under_memory_pressure()
for TCP. Our intent is to use it in fast paths.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
We plan to use sk_forced_wmem_schedule() in input path as well,
so make it non static and rename it.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Diagnosing problems related to Window Probes has been hard because
we lack a counter.
TCPWinProbe counts the number of ACK packets a sender has to send
at regular intervals to make sure a reverse ACK packet opening back
a window had not been lost.
TCPKeepAlive counts the number of ACK packets sent to keep TCP
flows alive (SO_KEEPALIVE)
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
With the advent of small rto timers in datacenter TCP,
(ip route ... rto_min x), the following can happen :
1) Qdisc is full, transmit fails.
TCP sets a timer based on icsk_rto to retry the transmit, without
exponential backoff.
With low icsk_rto, and lot of sockets, all cpus are servicing timer
interrupts like crazy.
Intent of the code was to retry with a timer between 200 (TCP_RTO_MIN)
and 500ms (TCP_RESOURCE_PROBE_INTERVAL)
2) Receivers can send zero windows if they don't drain their receive queue.
TCP sends zero window probes, based on icsk_rto current value, with
exponential backoff.
With /proc/sys/net/ipv4/tcp_retries2 being 15 (or even smaller in
some cases), sender can abort in less than one or two minutes !
If receiver stops the sender, it obviously doesn't care of very tight
rto. Probability of dropping the ACK reopening the window is not
worth the risk.
Lets change the base timer to be at least 200ms (TCP_RTO_MIN) for these
events (but not normal RTO based retransmits)
A followup patch adds a new SNMP counter, as it would have helped a lot
diagnosing this issue.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Presence of an unbound loop in tcp_send_fin() had always been hard
to explain when analyzing crash dumps involving gigantic dying processes
with millions of sockets.
Lets try a different strategy :
In case of memory pressure, try to add the FIN flag to last packet
in write queue, even if packet was already sent. TCP stack will
be able to deliver this FIN after a timeout event. Note that this
FIN being delivered by a retransmit, it also carries a Push flag
given our current implementation.
By checking sk_under_memory_pressure(), we anticipate that cooking
many FIN packets might deplete tcp memory.
In the case we could not allocate a packet, even with __GFP_WAIT
allocation, then not sending a FIN seems quite reasonable if it allows
to get rid of this socket, free memory, and not block the process from
eventually doing other useful work.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Using sk_stream_alloc_skb() in tcp_send_fin() is dangerous in
case a huge process is killed by OOM, and tcp_mem[2] is hit.
To be able to free memory we need to make progress, so this
patch allows FIN packets to not care about tcp_mem[2], if
skb allocation succeeded.
In a follow-up patch, we might abort tcp_send_fin() infinite loop
in case TIF_MEMDIE is set on this thread, as memory allocator
did its best getting extra memory already.
This patch reverts d22e153718 ("tcp: fix tcp fin memory accounting")
Fixes: d22e153718 ("tcp: fix tcp fin memory accounting")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
The dwmac-socfpga.c conflict was a case of a bug fix overlapping
changes in net-next to handle an error pointer differently.
Signed-off-by: David S. Miller <davem@davemloft.net>
I noticed tcpdump was giving funky timestamps for locally
generated SYNACK messages on loopback interface.
11:42:46.938990 IP 127.0.0.1.48245 > 127.0.0.2.23850: S
945476042:945476042(0) win 43690 <mss 65495,nop,nop,sackOK,nop,wscale 7>
20:28:58.502209 IP 127.0.0.2.23850 > 127.0.0.1.48245: S
3160535375:3160535375(0) ack 945476043 win 43690 <mss
65495,nop,nop,sackOK,nop,wscale 7>
This is because we need to clear skb->tstamp before
entering lower stack, otherwise net_timestamp_check()
does not set skb->tstamp.
Fixes: 7faee5c0d5 ("tcp: remove TCP_SKB_CB(skb)->when")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David 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: Daniel Lee <Longinus00@gmail.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David 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: Daniel Lee <Longinus00@gmail.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David 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: Ian Morris <ipm@chirality.org.uk>
Signed-off-by: David 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: Ian Morris <ipm@chirality.org.uk>
Signed-off-by: David S. Miller <davem@davemloft.net>
With request socks convergence, we no longer need
different lookup methods. A request socket can
use generic lookup function.
Add const qualifier to 2nd tcp_v[46]_md5_lookup() parameter.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Since request and established sockets now have same base,
there is no need to pass two pointers to tcp_v4_md5_hash_skb()
or tcp_v6_md5_hash_skb()
Also add a const qualifier to their struct tcp_md5sig_key argument.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
While timer handler effectively runs a rcu read locked section,
there is no explicit rcu_read_lock()/rcu_read_unlock() annotations
and lockdep can be confused here :
net/ipv4/tcp_ipv4.c-906- /* caller either holds rcu_read_lock() or socket lock */
net/ipv4/tcp_ipv4.c:907: md5sig = rcu_dereference_check(tp->md5sig_info,
net/ipv4/tcp_ipv4.c-908- sock_owned_by_user(sk) ||
net/ipv4/tcp_ipv4.c-909- lockdep_is_held(&sk->sk_lock.slock));
Let's explicitely acquire rcu_read_lock() in tcp_make_synack()
Before commit fa76ce7328 ("inet: get rid of central tcp/dccp listener
timer"), we were holding listener lock so lockdep was happy.
Fixes: fa76ce7328 ("inet: get rid of central tcp/dccp listener timer")
Signed-off-by: Eric DUmazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This reverts commit ca10b9e9a8.
No longer needed after commit eb8895debe
("tcp: tcp_make_synack() should use sock_wmalloc")
When under SYNFLOOD, we build lot of SYNACK and hit false sharing
because of multiple modifications done on sk_listener->sk_wmem_alloc
Since tcp_make_synack() uses sock_wmalloc(), there is no need
to call skb_set_owner_w() again, as this adds two atomic operations.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
drivers/net/ethernet/emulex/benet/be_main.c
net/core/sysctl_net_core.c
net/ipv4/inet_diag.c
The be_main.c conflict resolution was really tricky. The conflict
hunks generated by GIT were very unhelpful, to say the least. It
split functions in half and moved them around, when the real actual
conflict only existed solely inside of one function, that being
be_map_pci_bars().
So instead, to resolve this, I checked out be_main.c from the top
of net-next, then I applied the be_main.c changes from 'net' since
the last time I merged. And this worked beautifully.
The inet_diag.c and sysctl_net_core.c conflicts were simple
overlapping changes, and were easily to resolve.
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_send_fin() does not account for the memory it allocates properly, so
sk_forward_alloc can be negative in cases where we've sent a FIN:
ss example output (ss -amn | grep -B1 f4294):
tcp FIN-WAIT-1 0 1 192.168.0.1:45520 192.0.2.1:8080
skmem:(r0,rb87380,t0,tb87380,f4294966016,w1280,o0,bl0)
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
As per RFC4821 7.3. Selecting Probe Size, a probe timer should
be armed once probing has converged. Once this timer expired,
probing again to take advantage of any path PMTU change. The
recommended probing interval is 10 minutes per RFC1981. Probing
interval could be sysctled by sysctl_tcp_probe_interval.
Eric Dumazet suggested to implement pseudo timer based on 32bits
jiffies tcp_time_stamp instead of using classic timer for such
rare event.
Signed-off-by: Fan Du <fan.du@intel.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Current probe_size is chosen by doubling mss_cache,
the probing process will end shortly with a sub-optimal
mss size, and the link mtu will not be taken full
advantage of, in return, this will make user to tweak
tcp_base_mss with care.
Use binary search to choose probe_size in a fine
granularity manner, an optimal mss will be found
to boost performance as its maxmium.
In addition, introduce a sysctl_tcp_probe_threshold
to control when probing will stop in respect to
the width of search range.
Test env:
Docker instance with vxlan encapuslation(82599EB)
iperf -c 10.0.0.24 -t 60
before this patch:
1.26 Gbits/sec
After this patch: increase 26%
1.59 Gbits/sec
Signed-off-by: Fan Du <fan.du@intel.com>
Acked-by: John Heffner <johnwheffner@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Another TCP issue is triggered by ECN.
Under pressure, receiver gets ECN marks, and send back ACK packets
with ECE TCP flag. Senders enter CA_CWR state.
In this state, tcp_tso_should_defer() is short cut :
if (icsk->icsk_ca_state != TCP_CA_Open)
goto send_now;
This means that about all ACK packets we receive are triggering
a partial send, and because cwnd is kept small, we can only send
a small amount of data for each incoming ACK,
which in return generate more ACK packets.
Allowing CA_Open and CA_CWR states to enable TSO defer in
tcp_tso_should_defer() brings performance back :
TSO autodefer has more chance to defer under pressure.
This patch increases TSO and LRO/GRO efficiency back to normal levels,
and does not impact overall ECN behavior.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
With sysctl_tcp_min_tso_segs being 4, it is very possible
that tcp_tso_should_defer() decides not sending last 2 MSS
of initial window of 10 packets. This also applies if
autosizing decides to send X MSS per GSO packet, and cwnd
is not a multiple of X.
This patch implements an heuristic based on age of first
skb in write queue : If it was sent very recently (less than half srtt),
we can predict that no ACK packet will come in less than half rtt,
so deferring might cause an under utilization of our window.
This is visible on initial send (IW10) on web servers,
but more generally on some RPC, as the last part of the message
might need an extra RTT to get delivered.
Tested:
Ran following packetdrill test
// A simple server-side test that sends exactly an initial window (IW10)
// worth of packets.
`sysctl -e -q net.ipv4.tcp_min_tso_segs=4`
0.000 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
+.1 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7>
+0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6>
+.1 < . 1:1(0) ack 1 win 257
+0 accept(3, ..., ...) = 4
+0 write(4, ..., 14600) = 14600
+0 > . 1:5841(5840) ack 1 win 457
+0 > . 5841:11681(5840) ack 1 win 457
// Following packet should be sent right now.
+0 > P. 11681:14601(2920) ack 1 win 457
+.1 < . 1:1(0) ack 14601 win 257
+0 close(4) = 0
+0 > F. 14601:14601(0) ack 1
+.1 < F. 1:1(0) ack 14602 win 257
+0 > . 14602:14602(0) ack 2
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
TSO relies on ability to defer sending a small amount of packets.
Heuristic is to wait for future ACKS in hope to send more packets at once.
Current algorithm uses a per socket tso_deferred field as a pseudo timer.
This pseudo timer relies on future ACK, but there is no guarantee
we receive them in time.
Fix would be to use a real timer, but cost of such timer is probably too
expensive for typical cases.
This patch changes the logic to test the time of last transmit,
because we should not add bursts of more than 1ms for any given flow.
We've used this patch for about two years at Google, before FQ/pacing
as it would reduce a fair amount of bursts.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Packetization Layer Path MTU Discovery works separately beside
Path MTU Discovery at IP level, different net namespace has
various requirements on which one to chose, e.g., a virutalized
container instance would require TCP PMTU to probe an usable
effective mtu for underlying tunnel, while the host would
employ classical ICMP based PMTU to function.
Hence making TCP PMTU mechanism per net namespace to decouple
two functionality. Furthermore the probe base MSS should also
be configured separately for each namespace.
Signed-off-by: Fan Du <fan.du@intel.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
When we added pacing to TCP, we decided to let sch_fq take care
of actual pacing.
All TCP had to do was to compute sk->pacing_rate using simple formula:
sk->pacing_rate = 2 * cwnd * mss / rtt
It works well for senders (bulk flows), but not very well for receivers
or even RPC :
cwnd on the receiver can be less than 10, rtt can be around 100ms, so we
can end up pacing ACK packets, slowing down the sender.
Really, only the sender should pace, according to its own logic.
Instead of adding a new bit in skb, or call yet another flow
dissection, we tweak skb->truesize to a small value (2), and
we instruct sch_fq to use new helper and not pace pure ack.
Note this also helps TCP small queue, as ack packets present
in qdisc/NIC do not prevent sending a data packet (RPC workload)
This helps to reduce tx completion overhead, ack packets can use regular
sock_wfree() instead of tcp_wfree() which is a bit more expensive.
This has no impact in the case packets are sent to loopback interface,
as we do not coalesce ack packets (were we would detect skb->truesize
lie)
In case netem (with a delay) is used, skb_orphan_partial() also sets
skb->truesize to 1.
This patch is a combination of two patches we used for about one year at
Google.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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: Al Viro <viro@zeniv.linux.org.uk>
This work adds the possibility to define a per route/destination
congestion control algorithm. Generally, this opens up the possibility
for a machine with different links to enforce specific congestion
control algorithms with optimal strategies for each of them based
on their network characteristics, even transparently for a single
application listening on all links.
For our specific use case, this additionally facilitates deployment
of DCTCP, for example, applications can easily serve internal
traffic/dsts in DCTCP and external one with CUBIC. Other scenarios
would also allow for utilizing e.g. long living, low priority
background flows for certain destinations/routes while still being
able for normal traffic to utilize the default congestion control
algorithm. We also thought about a per netns setting (where different
defaults are possible), but given its actually a link specific
property, we argue that a per route/destination setting is the most
natural and flexible.
The administrator can utilize this through ip-route(8) by appending
"congctl [lock] <name>", where <name> denotes the name of a
congestion control algorithm and the optional lock parameter allows
to enforce the given algorithm so that applications in user space
would not be allowed to overwrite that algorithm for that destination.
The dst metric lookups are being done when a dst entry is already
available in order to avoid a costly lookup and still before the
algorithms are being initialized, thus overhead is very low when the
feature is not being used. While the client side would need to drop
the current reference on the module, on server side this can actually
even be avoided as we just got a flat-copied socket clone.
Joint work with Florian Westphal.
Suggested-by: Hannes Frederic Sowa <hannes@stressinduktion.org>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Thomas Jarosch reported IPsec TCP stalls when a PMTU event occurs.
In fact the problem was completely unrelated to IPsec. The bug is
also reproducible if you just disable TSO/GSO.
The problem is that when the MSS goes down, existing queued packet
on the TX queue that have not been transmitted yet all look like
TSO packets and get treated as such.
This then triggers a bug where tcp_mss_split_point tells us to
generate a zero-sized packet on the TX queue. Once that happens
we're screwed because the zero-sized packet can never be removed
by ACKs.
Fixes: 1485348d24 ("tcp: Apply device TSO segment limit earlier")
Reported-by: Thomas Jarosch <thomas.jarosch@intra2net.com>
Signed-off-by: Herbert Xu <herbert@gondor.apana.org.au>
Cheers,
Signed-off-by: David S. Miller <davem@davemloft.net>
Commit 95bd09eb27 ("tcp: TSO packets automatic sizing") tried to
control TSO size, but did this at the wrong place (sendmsg() time)
At sendmsg() time, we might have a pessimistic view of flow rate,
and we end up building very small skbs (with 2 MSS per skb).
This is bad because :
- It sends small TSO packets even in Slow Start where rate quickly
increases.
- It tends to make socket write queue very big, increasing tcp_ack()
processing time, but also increasing memory needs, not necessarily
accounted for, as fast clones overhead is currently ignored.
- Lower GRO efficiency and more ACK packets.
Servers with a lot of small lived connections suffer from this.
Lets instead fill skbs as much as possible (64KB of payload), but split
them at xmit time, when we have a precise idea of the flow rate.
skb split is actually quite efficient.
Patch looks bigger than necessary, because TCP Small Queue decision now
has to take place after the eventual split.
As Neal suggested, introduce a new tcp_tso_autosize() helper, so that
tcp_tso_should_defer() can be synchronized on same goal.
Rename tp->xmit_size_goal_segs to tp->gso_segs, as this variable
contains number of mss that we can put in GSO packet, and is not
related to the autosizing goal anymore.
Tested:
40 ms rtt link
nstat >/dev/null
netperf -H remote -l -2000000 -- -s 1000000
nstat | egrep "IpInReceives|IpOutRequests|TcpOutSegs|IpExtOutOctets"
Before patch :
Recv Send Send
Socket Socket Message Elapsed
Size Size Size Time Throughput
bytes bytes bytes secs. 10^6bits/s
87380 2000000 2000000 0.36 44.22
IpInReceives 600 0.0
IpOutRequests 599 0.0
TcpOutSegs 1397 0.0
IpExtOutOctets 2033249 0.0
After patch :
Recv Send Send
Socket Socket Message Elapsed
Size Size Size Time Throughput
bytes bytes bytes secs. 10^6bits/sec
87380 2000000 2000000 0.36 44.27
IpInReceives 221 0.0
IpOutRequests 232 0.0
TcpOutSegs 1397 0.0
IpExtOutOctets 2013953 0.0
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Note that the code _using_ ->msg_iter at that point will be very
unhappy with anything other than unshifted iovec-backed iov_iter.
We still need to convert users to proper primitives.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
While working on sk_forward_alloc problems reported by Denys
Fedoryshchenko, we found that tcp connect() (and fastopen) do not call
sk_wmem_schedule() for SYN packet (and/or SYN/DATA packet), so
sk_forward_alloc is negative while connect is in progress.
We can fix this by calling regular sk_stream_alloc_skb() both for the
SYN packet (in tcp_connect()) and the syn_data packet in
tcp_send_syn_data()
Then, tcp_send_syn_data() can avoid copying syn_data as we simply
can manipulate syn_data->cb[] to remove SYN flag (and increment seq)
Instead of open coding memcpy_fromiovecend(), simply use this helper.
This leaves in socket write queue clean fast clone skbs.
This was tested against our fastopen packetdrill tests.
Reported-by: Denys Fedoryshchenko <nuclearcat@nuclearcat.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
In DC world, GSO packets initially cooked by tcp_sendmsg() are usually
big, as sk_pacing_rate is high.
When network is congested, cwnd can be smaller than the GSO packets
found in socket write queue. tcp_write_xmit() splits GSO packets
using the available cwnd, and we end up sending a single GSO packet,
consuming all available cwnd.
With GRO aggregation on the receiver, we might handle a single GRO
packet, sending back a single ACK.
1) This single ACK might be lost
TLP or RTO are forced to attempt a retransmit.
2) This ACK releases a full cwnd, sender sends another big GSO packet,
in a ping pong mode.
This behavior does not fill the pipes in the best way, because of
scheduling artifacts.
Make sure we always have at least two GSO packets in flight.
This allows us to safely increase GRO efficiency without risking
spurious retransmits.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David 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 255cac91c3 ("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/335797
Suggested-by: Hannes Frederic Sowa <hannes@stressinduktion.org>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: David S. Miller <davem@davemloft.net>
Some drivers are unable to perform TX completions in a bound time.
They instead call skb_orphan()
Problem is skb_fclone_busy() has to detect this case, otherwise
we block TCP retransmits and can freeze unlucky tcp sessions on
mostly idle hosts.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Fixes: 1f3279ae0c ("tcp: avoid retransmits of TCP packets hanging in host queues")
Signed-off-by: David S. Miller <davem@davemloft.net>
Pull networking fixes from David Miller:
1) Include fixes for netrom and dsa (Fabian Frederick and Florian
Fainelli)
2) Fix FIXED_PHY support in stmmac, from Giuseppe CAVALLARO.
3) Several SKB use after free fixes (vxlan, openvswitch, vxlan,
ip_tunnel, fou), from Li ROngQing.
4) fec driver PTP support fixes from Luwei Zhou and Nimrod Andy.
5) Use after free in virtio_net, from Michael S Tsirkin.
6) Fix flow mask handling for megaflows in openvswitch, from Pravin B
Shelar.
7) ISDN gigaset and capi bug fixes from Tilman Schmidt.
8) Fix route leak in ip_send_unicast_reply(), from Vasily Averin.
9) Fix two eBPF JIT bugs on x86, from Alexei Starovoitov.
10) TCP_SKB_CB() reorganization caused a few regressions, fixed by Cong
Wang and Eric Dumazet.
11) Don't overwrite end of SKB when parsing malformed sctp ASCONF
chunks, from Daniel Borkmann.
12) Don't call sock_kfree_s() with NULL pointers, this function also has
the side effect of adjusting the socket memory usage. From Cong Wang.
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/net: (90 commits)
bna: fix skb->truesize underestimation
net: dsa: add includes for ethtool and phy_fixed definitions
openvswitch: Set flow-key members.
netrom: use linux/uaccess.h
dsa: Fix conversion from host device to mii bus
tipc: fix bug in bundled buffer reception
ipv6: introduce tcp_v6_iif()
sfc: add support for skb->xmit_more
r8152: return -EBUSY for runtime suspend
ipv4: fix a potential use after free in fou.c
ipv4: fix a potential use after free in ip_tunnel_core.c
hyperv: Add handling of IP header with option field in netvsc_set_hash()
openvswitch: Create right mask with disabled megaflows
vxlan: fix a free after use
openvswitch: fix a use after free
ipv4: dst_entry leak in ip_send_unicast_reply()
ipv4: clean up cookie_v4_check()
ipv4: share tcp_v4_save_options() with cookie_v4_check()
ipv4: call __ip_options_echo() in cookie_v4_check()
atm: simplify lanai.c by using module_pci_driver
...
Pull percpu consistent-ops changes from Tejun Heo:
"Way back, before the current percpu allocator was implemented, static
and dynamic percpu memory areas were allocated and handled separately
and had their own accessors. The distinction has been gone for many
years now; however, the now duplicate two sets of accessors remained
with the pointer based ones - this_cpu_*() - evolving various other
operations over time. During the process, we also accumulated other
inconsistent operations.
This pull request contains Christoph's patches to clean up the
duplicate accessor situation. __get_cpu_var() uses are replaced with
with this_cpu_ptr() and __this_cpu_ptr() with raw_cpu_ptr().
Unfortunately, the former sometimes is tricky thanks to C being a bit
messy with the distinction between lvalues and pointers, which led to
a rather ugly solution for cpumask_var_t involving the introduction of
this_cpu_cpumask_var_ptr().
This converts most of the uses but not all. Christoph will follow up
with the remaining conversions in this merge window and hopefully
remove the obsolete accessors"
* 'for-3.18-consistent-ops' of git://git.kernel.org/pub/scm/linux/kernel/git/tj/percpu: (38 commits)
irqchip: Properly fetch the per cpu offset
percpu: Resolve ambiguities in __get_cpu_var/cpumask_var_t -fix
ia64: sn_nodepda cannot be assigned to after this_cpu conversion. Use __this_cpu_write.
percpu: Resolve ambiguities in __get_cpu_var/cpumask_var_t
Revert "powerpc: Replace __get_cpu_var uses"
percpu: Remove __this_cpu_ptr
clocksource: Replace __this_cpu_ptr with raw_cpu_ptr
sparc: Replace __get_cpu_var uses
avr32: Replace __get_cpu_var with __this_cpu_write
blackfin: Replace __get_cpu_var uses
tile: Use this_cpu_ptr() for hardware counters
tile: Replace __get_cpu_var uses
powerpc: Replace __get_cpu_var uses
alpha: Replace __get_cpu_var
ia64: Replace __get_cpu_var uses
s390: cio driver &__get_cpu_var replacements
s390: Replace __get_cpu_var uses
mips: Replace __get_cpu_var uses
MIPS: Replace __get_cpu_var uses in FPU emulator.
arm: Replace __this_cpu_ptr with raw_cpu_ptr
...
TCP Small queues tries to keep number of packets in qdisc
as small as possible, and depends on a tasklet to feed following
packets at TX completion time.
Choice of tasklet was driven by latencies requirements.
Then, TCP stack tries to avoid reorders, by locking flows with
outstanding packets in qdisc in a given TX queue.
What can happen is that many flows get attracted by a low performing
TX queue, and cpu servicing TX completion has to feed packets for all of
them, making this cpu 100% busy in softirq mode.
This became particularly visible with latest skb->xmit_more support
Strategy adopted in this patch is to detect when tcp_wfree() is called
from ksoftirqd and let the outstanding queue for this flow being drained
before feeding additional packets, so that skb->ooo_okay can be set
to allow select_queue() to select the optimal queue :
Incoming ACKS are normally handled by different cpus, so this patch
gives more chance for these cpus to take over the burden of feeding
qdisc with future packets.
Tested:
lpaa23:~# ./super_netperf 1400 --google-pacing-rate 3028000 -H lpaa24 -l 3600 &
lpaa23:~# sar -n DEV 1 10 | grep eth1
06:16:18 AM eth1 595448.00 1190564.00 38381.09 1760253.12 0.00 0.00 1.00
06:16:19 AM eth1 594858.00 1189686.00 38340.76 1758952.72 0.00 0.00 0.00
06:16:20 AM eth1 597017.00 1194019.00 38480.79 1765370.29 0.00 0.00 1.00
06:16:21 AM eth1 595450.00 1190936.00 38380.19 1760805.05 0.00 0.00 0.00
06:16:22 AM eth1 596385.00 1193096.00 38442.56 1763976.29 0.00 0.00 1.00
06:16:23 AM eth1 598155.00 1195978.00 38552.97 1768264.60 0.00 0.00 0.00
06:16:24 AM eth1 594405.00 1188643.00 38312.57 1757414.89 0.00 0.00 1.00
06:16:25 AM eth1 593366.00 1187154.00 38252.16 1755195.83 0.00 0.00 0.00
06:16:26 AM eth1 593188.00 1186118.00 38232.88 1753682.57 0.00 0.00 1.00
06:16:27 AM eth1 596301.00 1192241.00 38440.94 1762733.09 0.00 0.00 0.00
Average: eth1 595457.30 1190843.50 38381.69 1760664.84 0.00 0.00 0.50
lpaa23:~# ./tc -s -d qd sh dev eth1 | grep backlog
backlog 7606336b 2513p requeues 167982
backlog 224072b 74p requeues 566
backlog 581376b 192p requeues 5598
backlog 181680b 60p requeues 1070
backlog 5305056b 1753p requeues 110166 // Here, this TX queue is attracting flows
backlog 157456b 52p requeues 1758
backlog 672216b 222p requeues 3025
backlog 60560b 20p requeues 24541
backlog 448144b 148p requeues 21258
lpaa23:~# echo 1 >/proc/sys/net/ipv4/tcp_tsq_enable_tcp_wfree_ksoftirqd_detect
Immediate jump to full bandwidth, and traffic is properly
shard on all tx queues.
lpaa23:~# sar -n DEV 1 10 | grep eth1
06:16:46 AM eth1 1397632.00 2795397.00 90081.87 4133031.26 0.00 0.00 1.00
06:16:47 AM eth1 1396874.00 2793614.00 90032.99 4130385.46 0.00 0.00 0.00
06:16:48 AM eth1 1395842.00 2791600.00 89966.46 4127409.67 0.00 0.00 1.00
06:16:49 AM eth1 1395528.00 2791017.00 89946.17 4126551.24 0.00 0.00 0.00
06:16:50 AM eth1 1397891.00 2795716.00 90098.74 4133497.39 0.00 0.00 1.00
06:16:51 AM eth1 1394951.00 2789984.00 89908.96 4125022.51 0.00 0.00 0.00
06:16:52 AM eth1 1394608.00 2789190.00 89886.90 4123851.36 0.00 0.00 1.00
06:16:53 AM eth1 1395314.00 2790653.00 89934.33 4125983.09 0.00 0.00 0.00
06:16:54 AM eth1 1396115.00 2792276.00 89984.25 4128411.21 0.00 0.00 1.00
06:16:55 AM eth1 1396829.00 2793523.00 90030.19 4130250.28 0.00 0.00 0.00
Average: eth1 1396158.40 2792297.00 89987.09 4128439.35 0.00 0.00 0.50
lpaa23:~# tc -s -d qd sh dev eth1 | grep backlog
backlog 7900052b 2609p requeues 173287
backlog 878120b 290p requeues 589
backlog 1068884b 354p requeues 5621
backlog 996212b 329p requeues 1088
backlog 984100b 325p requeues 115316
backlog 956848b 316p requeues 1781
backlog 1080996b 357p requeues 3047
backlog 975016b 322p requeues 24571
backlog 990156b 327p requeues 21274
(All 8 TX queues get a fair share of the traffic)
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
TCP Small Queues (tcp_tsq_handler()) can hold one reference on
sk->sk_wmem_alloc, preventing skb->ooo_okay being set.
We should relax test done to set skb->ooo_okay to take care
of this extra reference.
Minimal truesize of skb containing one byte of payload is
SKB_TRUESIZE(1)
Without this fix, we have more chance locking flows into the wrong
transmit queue.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Lets use a proper structure to clearly document and implement
skb fast clones.
Then, we might experiment more easily alternative layouts.
This patch adds a new skb_fclone_busy() helper, used by tcp and xfrm,
to stop leaking of implementation details.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Suggested by Stephen. Also drop inline keyword and let compiler decide.
gcc 4.7.3 decides to no longer inline tcp_ecn_check_ce, so split it up.
The actual evaluation is not inlined anymore while the ECN_OK test is.
Suggested-by: Stephen Hemminger <stephen@networkplumber.org>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: David S. Miller <davem@davemloft.net>
This work adds the DataCenter TCP (DCTCP) congestion control
algorithm [1], which has been first published at SIGCOMM 2010 [2],
resp. follow-up analysis at SIGMETRICS 2011 [3] (and also, more
recently as an informational IETF draft available at [4]).
DCTCP is an enhancement to the TCP congestion control algorithm for
data center networks. Typical data center workloads are i.e.
i) partition/aggregate (queries; bursty, delay sensitive), ii) short
messages e.g. 50KB-1MB (for coordination and control state; delay
sensitive), and iii) large flows e.g. 1MB-100MB (data update;
throughput sensitive). DCTCP has therefore been designed for such
environments to provide/achieve the following three requirements:
* High burst tolerance (incast due to partition/aggregate)
* Low latency (short flows, queries)
* High throughput (continuous data updates, large file
transfers) with commodity, shallow buffered switches
The basic idea of its design consists of two fundamentals: i) on the
switch side, packets are being marked when its internal queue
length > threshold K (K is chosen so that a large enough headroom
for marked traffic is still available in the switch queue); ii) the
sender/host side maintains a moving average of the fraction of marked
packets, so each RTT, F is being updated as follows:
F := X / Y, where X is # of marked ACKs, Y is total # of ACKs
alpha := (1 - g) * alpha + g * F, where g is a smoothing constant
The resulting alpha (iow: probability that switch queue is congested)
is then being used in order to adaptively decrease the congestion
window W:
W := (1 - (alpha / 2)) * W
The means for receiving marked packets resp. marking them on switch
side in DCTCP is the use of ECN.
RFC3168 describes a mechanism for using Explicit Congestion Notification
from the switch for early detection of congestion, rather than waiting
for segment loss to occur.
However, this method only detects the presence of congestion, not
the *extent*. In the presence of mild congestion, it reduces the TCP
congestion window too aggressively and unnecessarily affects the
throughput of long flows [4].
DCTCP, as mentioned, enhances Explicit Congestion Notification (ECN)
processing to estimate the fraction of bytes that encounter congestion,
rather than simply detecting that some congestion has occurred. DCTCP
then scales the TCP congestion window based on this estimate [4],
thus it can derive multibit feedback from the information present in
the single-bit sequence of marks in its control law. And thus act in
*proportion* to the extent of congestion, not its *presence*.
Switches therefore set the Congestion Experienced (CE) codepoint in
packets when internal queue lengths exceed threshold K. Resulting,
DCTCP delivers the same or better throughput than normal TCP, while
using 90% less buffer space.
It was found in [2] that DCTCP enables the applications to handle 10x
the current background traffic, without impacting foreground traffic.
Moreover, a 10x increase in foreground traffic did not cause any
timeouts, and thus largely eliminates TCP incast collapse problems.
The algorithm itself has already seen deployments in large production
data centers since then.
We did a long-term stress-test and analysis in a data center, short
summary of our TCP incast tests with iperf compared to cubic:
This test measured DCTCP throughput and latency and compared it with
CUBIC throughput and latency for an incast scenario. In this test, 19
senders sent at maximum rate to a single receiver. The receiver simply
ran iperf -s.
The senders ran iperf -c <receiver> -t 30. All senders started
simultaneously (using local clocks synchronized by ntp).
This test was repeated multiple times. Below shows the results from a
single test. Other tests are similar. (DCTCP results were extremely
consistent, CUBIC results show some variance induced by the TCP timeouts
that CUBIC encountered.)
For this test, we report statistics on the number of TCP timeouts,
flow throughput, and traffic latency.
1) Timeouts (total over all flows, and per flow summaries):
CUBIC DCTCP
Total 3227 25
Mean 169.842 1.316
Median 183 1
Max 207 5
Min 123 0
Stddev 28.991 1.600
Timeout data is taken by measuring the net change in netstat -s
"other TCP timeouts" reported. As a result, the timeout measurements
above are not restricted to the test traffic, and we believe that it
is likely that all of the "DCTCP timeouts" are actually timeouts for
non-test traffic. We report them nevertheless. CUBIC will also include
some non-test timeouts, but they are drawfed by bona fide test traffic
timeouts for CUBIC. Clearly DCTCP does an excellent job of preventing
TCP timeouts. DCTCP reduces timeouts by at least two orders of
magnitude and may well have eliminated them in this scenario.
2) Throughput (per flow in Mbps):
CUBIC DCTCP
Mean 521.684 521.895
Median 464 523
Max 776 527
Min 403 519
Stddev 105.891 2.601
Fairness 0.962 0.999
Throughput data was simply the average throughput for each flow
reported by iperf. By avoiding TCP timeouts, DCTCP is able to
achieve much better per-flow results. In CUBIC, many flows
experience TCP timeouts which makes flow throughput unpredictable and
unfair. DCTCP, on the other hand, provides very clean predictable
throughput without incurring TCP timeouts. Thus, the standard deviation
of CUBIC throughput is dramatically higher than the standard deviation
of DCTCP throughput.
Mean throughput is nearly identical because even though cubic flows
suffer TCP timeouts, other flows will step in and fill the unused
bandwidth. Note that this test is something of a best case scenario
for incast under CUBIC: it allows other flows to fill in for flows
experiencing a timeout. Under situations where the receiver is issuing
requests and then waiting for all flows to complete, flows cannot fill
in for timed out flows and throughput will drop dramatically.
3) Latency (in ms):
CUBIC DCTCP
Mean 4.0088 0.04219
Median 4.055 0.0395
Max 4.2 0.085
Min 3.32 0.028
Stddev 0.1666 0.01064
Latency for each protocol was computed by running "ping -i 0.2
<receiver>" from a single sender to the receiver during the incast
test. For DCTCP, "ping -Q 0x6 -i 0.2 <receiver>" was used to ensure
that traffic traversed the DCTCP queue and was not dropped when the
queue size was greater than the marking threshold. The summary
statistics above are over all ping metrics measured between the single
sender, receiver pair.
The latency results for this test show a dramatic difference between
CUBIC and DCTCP. CUBIC intentionally overflows the switch buffer
which incurs the maximum queue latency (more buffer memory will lead
to high latency.) DCTCP, on the other hand, deliberately attempts to
keep queue occupancy low. The result is a two orders of magnitude
reduction of latency with DCTCP - even with a switch with relatively
little RAM. Switches with larger amounts of RAM will incur increasing
amounts of latency for CUBIC, but not for DCTCP.
4) Convergence and stability test:
This test measured the time that DCTCP took to fairly redistribute
bandwidth when a new flow commences. It also measured DCTCP's ability
to remain stable at a fair bandwidth distribution. DCTCP is compared
with CUBIC for this test.
At the commencement of this test, a single flow is sending at maximum
rate (near 10 Gbps) to a single receiver. One second after that first
flow commences, a new flow from a distinct server begins sending to
the same receiver as the first flow. After the second flow has sent
data for 10 seconds, the second flow is terminated. The first flow
sends for an additional second. Ideally, the bandwidth would be evenly
shared as soon as the second flow starts, and recover as soon as it
stops.
The results of this test are shown below. Note that the flow bandwidth
for the two flows was measured near the same time, but not
simultaneously.
DCTCP performs nearly perfectly within the measurement limitations
of this test: bandwidth is quickly distributed fairly between the two
flows, remains stable throughout the duration of the test, and
recovers quickly. CUBIC, in contrast, is slow to divide the bandwidth
fairly, and has trouble remaining stable.
CUBIC DCTCP
Seconds Flow 1 Flow 2 Seconds Flow 1 Flow 2
0 9.93 0 0 9.92 0
0.5 9.87 0 0.5 9.86 0
1 8.73 2.25 1 6.46 4.88
1.5 7.29 2.8 1.5 4.9 4.99
2 6.96 3.1 2 4.92 4.94
2.5 6.67 3.34 2.5 4.93 5
3 6.39 3.57 3 4.92 4.99
3.5 6.24 3.75 3.5 4.94 4.74
4 6 3.94 4 5.34 4.71
4.5 5.88 4.09 4.5 4.99 4.97
5 5.27 4.98 5 4.83 5.01
5.5 4.93 5.04 5.5 4.89 4.99
6 4.9 4.99 6 4.92 5.04
6.5 4.93 5.1 6.5 4.91 4.97
7 4.28 5.8 7 4.97 4.97
7.5 4.62 4.91 7.5 4.99 4.82
8 5.05 4.45 8 5.16 4.76
8.5 5.93 4.09 8.5 4.94 4.98
9 5.73 4.2 9 4.92 5.02
9.5 5.62 4.32 9.5 4.87 5.03
10 6.12 3.2 10 4.91 5.01
10.5 6.91 3.11 10.5 4.87 5.04
11 8.48 0 11 8.49 4.94
11.5 9.87 0 11.5 9.9 0
SYN/ACK ECT test:
This test demonstrates the importance of ECT on SYN and SYN-ACK packets
by measuring the connection probability in the presence of competing
flows for a DCTCP connection attempt *without* ECT in the SYN packet.
The test was repeated five times for each number of competing flows.
Competing Flows 1 | 2 | 4 | 8 | 16
------------------------------
Mean Connection Probability 1 | 0.67 | 0.45 | 0.28 | 0
Median Connection Probability 1 | 0.65 | 0.45 | 0.25 | 0
As the number of competing flows moves beyond 1, the connection
probability drops rapidly.
Enabling DCTCP with this patch requires the following steps:
DCTCP must be running both on the sender and receiver side in your
data center, i.e.:
sysctl -w net.ipv4.tcp_congestion_control=dctcp
Also, ECN functionality must be enabled on all switches in your
data center for DCTCP to work. The default ECN marking threshold (K)
heuristic on the switch for DCTCP is e.g., 20 packets (30KB) at
1Gbps, and 65 packets (~100KB) at 10Gbps (K > 1/7 * C * RTT, [4]).
In above tests, for each switch port, traffic was segregated into two
queues. For any packet with a DSCP of 0x01 - or equivalently a TOS of
0x04 - the packet was placed into the DCTCP queue. All other packets
were placed into the default drop-tail queue. For the DCTCP queue,
RED/ECN marking was enabled, here, with a marking threshold of 75 KB.
More details however, we refer you to the paper [2] under section 3).
There are no code changes required to applications running in user
space. DCTCP has been implemented in full *isolation* of the rest of
the TCP code as its own congestion control module, so that it can run
without a need to expose code to the core of the TCP stack, and thus
nothing changes for non-DCTCP users.
Changes in the CA framework code are minimal, and DCTCP algorithm
operates on mechanisms that are already available in most Silicon.
The gain (dctcp_shift_g) is currently a fixed constant (1/16) from
the paper, but we leave the option that it can be chosen carefully
to a different value by the user.
In case DCTCP is being used and ECN support on peer site is off,
DCTCP falls back after 3WHS to operate in normal TCP Reno mode.
ss {-4,-6} -t -i diag interface:
... dctcp wscale:7,7 rto:203 rtt:2.349/0.026 mss:1448 cwnd:2054
ssthresh:1102 ce_state 0 alpha 15 ab_ecn 0 ab_tot 735584
send 10129.2Mbps pacing_rate 20254.1Mbps unacked:1822 retrans:0/15
reordering:101 rcv_space:29200
... dctcp-reno wscale:7,7 rto:201 rtt:0.711/1.327 ato:40 mss:1448
cwnd:10 ssthresh:1102 fallback_mode send 162.9Mbps pacing_rate
325.5Mbps rcv_rtt:1.5 rcv_space:29200
More information about DCTCP can be found in [1-4].
[1] http://simula.stanford.edu/~alizade/Site/DCTCP.html
[2] http://simula.stanford.edu/~alizade/Site/DCTCP_files/dctcp-final.pdf
[3] http://simula.stanford.edu/~alizade/Site/DCTCP_files/dctcp_analysis-full.pdf
[4] http://tools.ietf.org/html/draft-bensley-tcpm-dctcp-00
Joint work with Florian Westphal and Glenn Judd.
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Glenn Judd <glenn.judd@morganstanley.com>
Acked-by: Stephen Hemminger <stephen@networkplumber.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
DataCenter TCP (DCTCP) determines cwnd growth based on ECN information
and ACK properties, e.g. ACK that updates window is treated differently
than DUPACK.
Also DCTCP needs information whether ACK was delayed ACK. Furthermore,
DCTCP also implements a CE state machine that keeps track of CE markings
of incoming packets.
Therefore, extend the congestion control framework to provide these
event types, so that DCTCP can be properly implemented as a normal
congestion algorithm module outside of the core stack.
Joint work with Daniel Borkmann and Glenn Judd.
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: Glenn Judd <glenn.judd@morganstanley.com>
Acked-by: Stephen Hemminger <stephen@networkplumber.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch adds a flag to TCP congestion algorithms that allows
for requesting to mark IPv4/IPv6 sockets with transport as ECN
capable, that is, ECT(0), when required by a congestion algorithm.
It is currently used and needed in DataCenter TCP (DCTCP), as it
requires both peers to assert ECT on all IP packets sent - it
uses ECN feedback (i.e. CE, Congestion Encountered information)
from switches inside the data center to derive feedback to the
end hosts.
Therefore, simply add a new flag to icsk_ca_ops. Note that DCTCP's
algorithm/behaviour slightly diverges from RFC3168, therefore this
is only (!) enabled iff the assigned congestion control ops module
has requested this. By that, we can tightly couple this logic really
only to the provided congestion control ops.
Joint work with Florian Westphal and Glenn Judd.
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Glenn Judd <glenn.judd@morganstanley.com>
Acked-by: Stephen Hemminger <stephen@networkplumber.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
Our goal is to access no more than one cache line access per skb in
a write or receive queue when doing the various walks.
After recent TCP_SKB_CB() reorganizations, it is almost done.
Last part is tcp_skb_pcount() which currently uses
skb_shinfo(skb)->gso_segs, which is a terrible choice, because it needs
3 cache lines in current kernel (skb->head, skb->end, and
shinfo->gso_segs are all in 3 different cache lines, far from skb->cb)
This very simple patch reuses space currently taken by tcp_tw_isn
only in input path, as tcp_skb_pcount is only needed for skb stored in
write queue.
This considerably speeds up tcp_ack(), granted we avoid shinfo->tx_flags
to get SKBTX_ACK_TSTAMP, which seems possible.
This also speeds up all sack processing in general.
This speeds up tcp_sendmsg() because it no longer has to access/dirty
shinfo.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
TCP maintains lists of skb in write queue, and in receive queues
(in order and out of order queues)
Scanning these lists both in input and output path usually requires
access to skb->next, TCP_SKB_CB(skb)->seq, and TCP_SKB_CB(skb)->end_seq
These fields are currently in two different cache lines, meaning we
waste lot of memory bandwidth when these queues are big and flows
have either packet drops or packet reorders.
We can move TCP_SKB_CB(skb)->header at the end of TCP_SKB_CB, because
this header is not used in fast path. This allows TCP to search much faster
in the skb lists.
Even with regular flows, we save one cache line miss in fast path.
Thanks to Christoph Paasch for noticing we need to cleanup
skb->cb[] (IPCB/IP6CB) before entering IP stack in tx path,
and that I forgot IPCB use in tcp_v4_hnd_req() and tcp_v4_save_options().
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
While profiling TCP stack, I noticed one useless atomic operation
in tcp_sendmsg(), caused by skb_header_release().
It turns out all current skb_header_release() users have a fresh skb,
that no other user can see, so we can avoid one atomic operation.
Introduce __skb_header_release() to clearly document this.
This gave me a 1.5 % improvement on TCP_RR workload.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
icsk_rto is a 32bit field, and icsk_backoff can reach 15 by default,
or more if some sysctl (eg tcp_retries2) are changed.
Better use 64bit to perform icsk_rto << icsk_backoff operations
As Joe Perches suggested, add a helper for this.
Yuchung spotted the tcp_v4_err() case.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
The TCP_SKB_CB(skb)->when field no longer exists as of recent change
7faee5c0d5 ("tcp: remove TCP_SKB_CB(skb)->when"). And in any case,
tcp_fragment() is called on already-transmitted packets from the
__tcp_retransmit_skb() call site, so copying timestamps of any kind
in this spot is quite sensible.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Reported-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
After commit 740b0f1841 ("tcp: switch rtt estimations to usec resolution"),
we no longer need to maintain timestamps in two different fields.
TCP_SKB_CB(skb)->when can be removed, as same information sits in skb_mstamp.stamp_jiffies
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Replace uses of get_cpu_var for address calculation through this_cpu_ptr.
Cc: netdev@vger.kernel.org
Cc: Eric Dumazet <edumazet@google.com>
Acked-by: David S. Miller <davem@davemloft.net>
Signed-off-by: Christoph Lameter <cl@linux.com>
Signed-off-by: Tejun Heo <tj@kernel.org>
Make sure we use the correct address-family-specific function for
handling MTU reductions from within tcp_release_cb().
Previously AF_INET6 sockets were incorrectly always using the IPv6
code path when sometimes they were handling IPv4 traffic and thus had
an IPv4 dst.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Diagnosed-by: Willem de Bruijn <willemb@google.com>
Fixes: 563d34d057 ("tcp: dont drop MTU reduction indications")
Reviewed-by: Hannes Frederic Sowa <hannes@stressinduktion.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
Bytestream timestamps are correlated with a single byte in the skbuff,
recorded in skb_shinfo(skb)->tskey. When fragmenting skbuffs, ensure
that the tskey is set for the fragment in which the tskey falls
(seqno <= tskey < end_seqno).
The original implementation did not address fragmentation in
tcp_fragment or tso_fragment. Add code to inspect the sequence numbers
and move both tskey and the relevant tx_flags if necessary.
Reported-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: Willem de Bruijn <willemb@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Since Yuchung's 9b44190dc1 (tcp: refactor F-RTO), tcp_enter_cwr is always
called with set_ssthresh = 1. Thus, we can remove this argument from
tcp_enter_cwr. Further, as we remove this one, tcp_init_cwnd_reduction
is then always called with set_ssthresh = true, and so we can get rid of
this argument as well.
Cc: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Christoph Paasch <christoph.paasch@uclouvain.be>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
The undo code assumes that, upon entering loss recovery, TCP
1) always retransmit something
2) the retransmission never fails locally (e.g., qdisc drop)
so undo_marker is set in tcp_enter_recovery() and undo_retrans is
incremented only when tcp_retransmit_skb() is successful.
When the assumption is broken because TCP's cwnd is too small to
retransmit or the retransmit fails locally. The next (DUP)ACK
would incorrectly revert the cwnd and the congestion state in
tcp_try_undo_dsack() or tcp_may_undo(). Subsequent (DUP)ACKs
may enter the recovery state. The sender repeatedly enter and
(incorrectly) exit recovery states if the retransmits continue to
fail locally while receiving (DUP)ACKs.
The fix is to initialize undo_retrans to -1 and start counting on
the first retransmission. Always increment undo_retrans even if the
retransmissions fail locally because they couldn't cause DSACKs to
undo the cwnd reduction.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
For a connected socket we can precompute the flow hash for setting
in skb->hash on output. This is a performance advantage over
calculating the skb->hash for every packet on the connection. The
computation is done using the common hash algorithm to be consistent
with computations done for packets of the connection in other states
where thers is no socket (e.g. time-wait, syn-recv, syn-cookies).
This patch adds sk_txhash to the sock structure. inet_set_txhash and
ip6_set_txhash functions are added which are called from points in
TCP and UDP where socket moves to established state.
skb_set_hash_from_sk is a function which sets skb->hash from the
sock txhash value. This is called in UDP and TCP transmit path when
transmitting within the context of a socket.
Tested: ran super_netperf with 200 TCP_RR streams over a vxlan
interface (in this case skb_get_hash called on every TX packet to
create a UDP source port).
Before fix:
95.02% CPU utilization
154/256/505 90/95/99% latencies
1.13042e+06 tps
Time in functions:
0.28% skb_flow_dissect
0.21% __skb_get_hash
After fix:
94.95% CPU utilization
156/254/485 90/95/99% latencies
1.15447e+06
Neither __skb_get_hash nor skb_flow_dissect appear in perf
Signed-off-by: Tom Herbert <therbert@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Pull networking updates from David Miller:
1) Seccomp BPF filters can now be JIT'd, from Alexei Starovoitov.
2) Multiqueue support in xen-netback and xen-netfront, from Andrew J
Benniston.
3) Allow tweaking of aggregation settings in cdc_ncm driver, from Bjørn
Mork.
4) BPF now has a "random" opcode, from Chema Gonzalez.
5) Add more BPF documentation and improve test framework, from Daniel
Borkmann.
6) Support TCP fastopen over ipv6, from Daniel Lee.
7) Add software TSO helper functions and use them to support software
TSO in mvneta and mv643xx_eth drivers. From Ezequiel Garcia.
8) Support software TSO in fec driver too, from Nimrod Andy.
9) Add Broadcom SYSTEMPORT driver, from Florian Fainelli.
10) Handle broadcasts more gracefully over macvlan when there are large
numbers of interfaces configured, from Herbert Xu.
11) Allow more control over fwmark used for non-socket based responses,
from Lorenzo Colitti.
12) Do TCP congestion window limiting based upon measurements, from Neal
Cardwell.
13) Support busy polling in SCTP, from Neal Horman.
14) Allow RSS key to be configured via ethtool, from Venkata Duvvuru.
15) Bridge promisc mode handling improvements from Vlad Yasevich.
16) Don't use inetpeer entries to implement ID generation any more, it
performs poorly, from Eric Dumazet.
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/net-next: (1522 commits)
rtnetlink: fix userspace API breakage for iproute2 < v3.9.0
tcp: fixing TLP's FIN recovery
net: fec: Add software TSO support
net: fec: Add Scatter/gather support
net: fec: Increase buffer descriptor entry number
net: fec: Factorize feature setting
net: fec: Enable IP header hardware checksum
net: fec: Factorize the .xmit transmit function
bridge: fix compile error when compiling without IPv6 support
bridge: fix smatch warning / potential null pointer dereference
via-rhine: fix full-duplex with autoneg disable
bnx2x: Enlarge the dorq threshold for VFs
bnx2x: Check for UNDI in uncommon branch
bnx2x: Fix 1G-baseT link
bnx2x: Fix link for KR with swapped polarity lane
sctp: Fix sk_ack_backlog wrap-around problem
net/core: Add VF link state control policy
net/fsl: xgmac_mdio is dependent on OF_MDIO
net/fsl: Make xgmac_mdio read error message useful
net_sched: drr: warn when qdisc is not work conserving
...
Fix to a problem observed when losing a FIN segment that does not
contain data. In such situations, TLP is unable to recover from
*any* tail loss and instead adds at least PTO ms to the
retransmission process, i.e., RTO = RTO + PTO.
Signed-off-by: Per Hurtig <per.hurtig@kau.se>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Nandita Dukkipati <nanditad@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_fragment can be called from process context (from tso_fragment).
Add a new gfp parameter to allow it to preserve atomic memory if
possible.
Signed-off-by: Octavian Purdila <octavian.purdila@intel.com>
Reviewed-by: Christoph Paasch <christoph.paasch@uclouvain.be>
Signed-off-by: David S. Miller <davem@davemloft.net>
Experience with the recent e114a710aa ("tcp: fix cwnd limited
checking to improve congestion control") has shown that there are
common cases where that commit can cause cwnd to be much larger than
necessary. This leads to TSO autosizing cooking skbs that are too
large, among other things.
The main problems seemed to be:
(1) That commit attempted to predict the future behavior of the
connection by looking at the write queue (if TSO or TSQ limit
sending). That prediction sometimes overestimated future outstanding
packets.
(2) That commit always allowed cwnd to grow to twice the number of
outstanding packets (even in congestion avoidance, where this is not
needed).
This commit improves both of these, by:
(1) Switching to a measurement-based approach where we explicitly
track the largest number of packets in flight during the past window
("max_packets_out"), and remember whether we were cwnd-limited at the
moment we finished sending that flight.
(2) Only allowing cwnd to grow to twice the number of outstanding
packets ("max_packets_out") in slow start. In congestion avoidance
mode we now only allow cwnd to grow if it was fully utilized.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
To avoid large code duplication in IPv6, we need to first simplify
the complicate SYN-ACK sending code in tcp_v4_conn_request().
To use tcp_v4(6)_send_synack() to send all SYN-ACKs, we need to
initialize the mini socket's receive window before trying to
create the child socket and/or building the SYN-ACK packet. So we move
that initialization from tcp_make_synack() to tcp_v4_conn_request()
as a new function tcp_openreq_init_req_rwin().
After this refactoring the SYN-ACK sending code is simpler and easier
to implement Fast Open for IPv6.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Daniel Lee <longinus00@gmail.com>
Signed-off-by: Jerry Chu <hkchu@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Consolidate various cookie checking and generation code to simplify
the fast open processing. The main goal is to reduce code duplication
in tcp_v4_conn_request() for IPv6 support.
Removes two experimental sysctl flags TFO_SERVER_ALWAYS and
TFO_SERVER_COOKIE_NOT_CHKD used primarily for developmental debugging
purposes.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Daniel Lee <longinus00@gmail.com>
Signed-off-by: Jerry Chu <hkchu@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
drivers/net/ethernet/altera/altera_sgdma.c
net/netlink/af_netlink.c
net/sched/cls_api.c
net/sched/sch_api.c
The netlink conflict dealt with moving to netlink_capable() and
netlink_ns_capable() in the 'net' tree vs. supporting 'tc' operations
in non-init namespaces. These were simple transformations from
netlink_capable to netlink_ns_capable.
The Altera driver conflict was simply code removal overlapping some
void pointer cast cleanups in net-next.
Signed-off-by: David S. Miller <davem@davemloft.net>
Commit e114a710aa ("tcp: fix cwnd limited checking to improve
congestion control") obsoleted in_flight parameter from
tcp_is_cwnd_limited() and its callers.
This patch does the removal as promised.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Yuchung discovered tcp_is_cwnd_limited() was returning false in
slow start phase even if the application filled the socket write queue.
All congestion modules take into account tcp_is_cwnd_limited()
before increasing cwnd, so this behavior limits slow start from
probing the bandwidth at full speed.
The problem is that even if write queue is full (aka we are _not_
application limited), cwnd can be under utilized if TSO should auto
defer or TCP Small queues decided to hold packets.
So the in_flight can be kept to smaller value, and we can get to the
point tcp_is_cwnd_limited() returns false.
With TCP Small Queues and FQ/pacing, this issue is more visible.
We fix this by having tcp_cwnd_validate(), which is supposed to track
such things, take into account unsent_segs, the number of segs that we
are not sending at the moment due to TSO or TSQ, but intend to send
real soon. Then when we are cwnd-limited, remember this fact while we
are processing the window of ACKs that comes back.
For example, suppose we have a brand new connection with cwnd=10; we
are in slow start, and we send a flight of 9 packets. By the time we
have received ACKs for all 9 packets we want our cwnd to be 18.
We implement this by setting tp->lsnd_pending to 9, and
considering ourselves to be cwnd-limited while cwnd is less than
twice tp->lsnd_pending (2*9 -> 18).
This makes tcp_is_cwnd_limited() more understandable, by removing
the GSO/TSO kludge, that tried to work around the issue.
Note the in_flight parameter can be removed in a followup cleanup
patch.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Both TLP and Fast Open call __tcp_retransmit_skb() instead of
tcp_retransmit_skb() to avoid changing tp->retrans_out.
This has the side effect of missing SNMP counters increments as well
as tcp_info tcpi_total_retrans updates.
Fix this by moving the stats increments of into __tcp_retransmit_skb()
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Nandita Dukkipati <nanditad@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
In commit 0e280af026 ("tcp: introduce TCPSpuriousRtxHostQueues SNMP
counter") we added a logic to detect when a packet was retransmitted
while the prior clone was still in a qdisc or driver queue.
We are now confident we can do better, and catch the problem before
we fragment a TSO packet before retransmit, or in TLP path.
This patch fully exploits the logic by simply canceling the spurious
retransmit.
Original packet is in a queue and will eventually leave the host.
This helps to avoid network collapses when some events make the RTO
estimations very wrong, particularly when dealing with huge number of
sockets with synchronized blast.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Make tcp_cwnd_application_limited() static and move it from tcp_input.c to
tcp_output.c
Signed-off-by: Weiping Pan <wpan@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
ip_queue_xmit() assumes the skb it has to transmit is attached to an
inet socket. Commit 31c70d5956 ("l2tp: keep original skb ownership")
changed l2tp to not change skb ownership and thus broke this assumption.
One fix is to add a new 'struct sock *sk' parameter to ip_queue_xmit(),
so that we do not assume skb->sk points to the socket used by l2tp
tunnel.
Fixes: 31c70d5956 ("l2tp: keep original skb ownership")
Reported-by: Zhan Jianyu <nasa4836@gmail.com>
Tested-by: Zhan Jianyu <nasa4836@gmail.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
There is no need to allocate 15 bytes in excess for a SYNACK packet,
as it contains no data, only headers.
SYNACK are always generated in softirq context, and contain a single
segment, we can use TCP_INC_STATS_BH()
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
After commit d4589926d7 (tcp: refine TSO splits), tcp_nagle_check() does
not use parameter mss_now anymore.
Signed-off-by: Weiping Pan <panweiping3@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
drivers/net/usb/r8152.c
drivers/net/xen-netback/netback.c
Both the r8152 and netback conflicts were simple overlapping
changes.
Signed-off-by: David S. Miller <davem@davemloft.net>
All skb in socket write queue should be properly timestamped.
In case of FastOpen, we special case the SYN+DATA 'message' as we
queue in socket wrote queue the two fallback skbs:
1) SYN message by itself.
2) DATA segment by itself.
We should make sure these skbs have proper timestamps.
Add a WARN_ON_ONCE() to eventually catch future violations.
Fixes: 740b0f1841 ("tcp: switch rtt estimations to usec resolution")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Usage of skb->tstamp should remain private to TCP stack
(only set on packets on write queue, not on cloned ones)
Otherwise, packets given to loopback interface with a non null tstamp
can confuse netif_rx() / net_timestamp_check()
Other possibility would be to clear tstamp in loopback_xmit(),
as done in skb_scrub_packet()
Fixes: 740b0f1841 ("tcp: switch rtt estimations to usec resolution")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reported-by: Willem de Bruijn <willemb@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Can be invoked from non-BH context.
Based upon a patch by Eric Dumazet.
Fixes: f19c29e3e3 ("tcp: snmp stats for Fast Open, SYN rtx, and data pkts")
Reported-by: Sergey Senozhatsky <sergey.senozhatsky@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
drivers/net/wireless/ath/ath9k/recv.c
drivers/net/wireless/mwifiex/pcie.c
net/ipv6/sit.c
The SIT driver conflict consists of a bug fix being done by hand
in 'net' (missing u64_stats_init()) whilst in 'net-next' a helper
was created (netdev_alloc_pcpu_stats()) which takes care of this.
The two wireless conflicts were overlapping changes.
Signed-off-by: David S. Miller <davem@davemloft.net>
Add the following snmp stats:
TCPFastOpenActiveFail: Fast Open attempts (SYN/data) failed beacuse
the remote does not accept it or the attempts timed out.
TCPSynRetrans: number of SYN and SYN/ACK retransmits to break down
retransmissions into SYN, fast-retransmits, timeout retransmits, etc.
TCPOrigDataSent: number of outgoing packets with original data (excluding
retransmission but including data-in-SYN). This counter is different from
TcpOutSegs because TcpOutSegs also tracks pure ACKs. TCPOrigDataSent is
more useful to track the TCP retransmission rate.
Change TCPFastOpenActive to track only successful Fast Opens to be symmetric to
TCPFastOpenPassive.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Lawrence Brakmo <brakmo@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
RTT may be bogus with tall loss probe (TLP) when a packet
is retransmitted and latter (s)acked without TCPCB_SACKED_RETRANS flag.
For example, TLP calls __tcp_retransmit_skb() instead of
tcp_retransmit_skb(). The skb timestamps are updated but the sacked
flag is not marked with TCPCB_SACKED_RETRANS. As a result we'll
get bogus RTT in tcp_clean_rtx_queue() or in tcp_sacktag_one() on
spurious retransmission.
The fix is to apply the sticky flag TCP_EVER_RETRANS to enforce Karn's
check on RTT sampling. However this will disable F-RTO if timeout occurs
after TLP, by resetting undo_marker in tcp_enter_loss(). We relax this
check to only if any pending retransmists are still in-flight.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Upcoming congestion controls for TCP require usec resolution for RTT
estimations. Millisecond resolution is simply not enough these days.
FQ/pacing in DC environments also require this change for finer control
and removal of bimodal behavior due to the current hack in
tcp_update_pacing_rate() for 'small rtt'
TCP_CONG_RTT_STAMP is no longer needed.
As Julian Anastasov pointed out, we need to keep user compatibility :
tcp_metrics used to export RTT and RTTVAR in msec resolution,
so we added RTT_US and RTTVAR_US. An iproute2 patch is needed
to use the new attributes if provided by the kernel.
In this example ss command displays a srtt of 32 usecs (10Gbit link)
lpk51:~# ./ss -i dst lpk52
Netid State Recv-Q Send-Q Local Address:Port Peer
Address:Port
tcp ESTAB 0 1 10.246.11.51:42959
10.246.11.52:64614
cubic wscale:6,6 rto:201 rtt:0.032/0.001 ato:40 mss:1448
cwnd:10 send
3620.0Mbps pacing_rate 7240.0Mbps unacked:1 rcv_rtt:993 rcv_space:29559
Updated iproute2 ip command displays :
lpk51:~# ./ip tcp_metrics | grep 10.246.11.52
10.246.11.52 age 561.914sec cwnd 10 rtt 274us rttvar 213us source
10.246.11.51
Old binary displays :
lpk51:~# ip tcp_metrics | grep 10.246.11.52
10.246.11.52 age 561.914sec cwnd 10 rtt 250us rttvar 125us source
10.246.11.51
With help from Julian Anastasov, Stephen Hemminger and Yuchung Cheng
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Cc: Stephen Hemminger <stephen@networkplumber.org>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Larry Brakmo <brakmo@google.com>
Cc: Julian Anastasov <ja@ssi.bg>
Signed-off-by: David S. Miller <davem@davemloft.net>
Three counters are added:
- one to track when we went from non-zero to zero window
- one to track the reverse
- one counter incremented when we want to announce zero window,
but can't because we would shrink current window.
Suggested-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Florian Westphal <fw@strlen.de>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
While LINUX_MIB_TCPSPURIOUS_RTX_HOSTQUEUES can only be incremented
in tcp_transmit_skb() from softirq (incoming message or timer
activation), it is better to use NET_INC_STATS() instead of
NET_INC_STATS_BH() as tcp_transmit_skb() can be called from process
context.
This will avoid copy/paste confusion when/if we want to add
other SNMP counters in tcp_transmit_skb()
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Hannes Frederic Sowa <hannes@stressinduktion.org>
Cc: Florian Westphal <fw@strlen.de>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch fixes two bugs in fastopen :
1) The tcp_sendmsg(..., @size) argument was ignored.
Code was relying on user not fooling the kernel with iovec mismatches
2) When MTU is about 64KB, tcp_send_syn_data() attempts order-5
allocations, which are likely to fail when memory gets fragmented.
Fixes: 783237e8da ("net-tcp: Fast Open client - sending SYN-data")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Tested-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Currently the kernel tries to announce a zero window when free_space
is below the current receiver mss estimate.
When a sender is transmitting small packets and reader consumes data
slowly (or not at all), receiver might be unable to shrink the receive
win because
a) we cannot withdraw already-commited receive window, and,
b) we have to round the current rwin up to a multiple of the wscale
factor, else we would shrink the current window.
This causes the receive buffer to fill up until the rmem limit is hit.
When this happens, we start dropping packets.
Moreover, tcp_clamp_window may continue to grow sk_rcvbuf towards rmem[2]
even if socket is not being read from.
As we cannot avoid the "current_win is rounded up to multiple of mss"
issue [we would violate a) above] at least try to prevent the receive buf
growth towards tcp_rmem[2] limit by attempting to move to zero-window
announcement when free_space becomes less than 1/16 of the current
allowed receive buffer maximum. If tcp_rmem[2] is large, this will
increase our chances to get a zero-window announcement out in time.
Reproducer:
On server:
$ nc -l -p 12345
<suspend it: CTRL-Z>
Client:
#!/usr/bin/env python
import socket
import time
sock = socket.socket()
sock.setsockopt(socket.IPPROTO_TCP, socket.TCP_NODELAY, 1)
sock.connect(("192.168.4.1", 12345));
while True:
sock.send('A' * 23)
time.sleep(0.005)
socket buffer on server-side will grow until tcp_rmem[2] is hit,
at which point the client rexmits data until -EDTIMEOUT:
tcp_data_queue invokes tcp_try_rmem_schedule which will call
tcp_prune_queue which calls tcp_clamp_window(). And that function will
grow sk->sk_rcvbuf up until it eventually hits tcp_rmem[2].
Thanks to Eric Dumazet for running regression tests.
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Tested-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: David S. Miller <davem@davemloft.net>
One of my pet coding style peeves is the practice of
adding extra return; at the end of function.
Kill several instances of this in network code.
I suppose some coccinelle wizardy could do this automatically.
Signed-off-by: Stephen Hemminger <stephen@networkplumber.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
Commit 46d3ceabd8 ("tcp: TCP Small Queues") introduced a possible
regression for applications using TCP_NODELAY.
If TCP session is throttled because of tsq, we should consult
tp->nonagle when TX completion is done and allow us to send additional
segment, especially if this segment is not a full MSS.
Otherwise this segment is sent after an RTO.
[edumazet] : Cooked the changelog, added another fix about testing
sk_wmem_alloc twice because TX completion can happen right before
setting TSQ_THROTTLED bit.
This problem is particularly visible with recent auto corking,
but might also be triggered with low tcp_limit_output_bytes
values or NIC drivers delaying TX completion by hundred of usec,
and very low rtt.
Thomas Glanzmann for example reported an iscsi regression, caused
by tcp auto corking making this bug quite visible.
Fixes: 46d3ceabd8 ("tcp: TCP Small Queues")
Signed-off-by: John Ogness <john.ogness@linutronix.de>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reported-by: Thomas Glanzmann <thomas@glanzmann.de>
Signed-off-by: David S. Miller <davem@davemloft.net>
TCP pacing depends on an accurate srtt estimation.
Current srtt estimation is using jiffie resolution,
and has an artificial offset of at least 1 ms, which can produce
slowdowns when FQ/pacing is used, especially in DC world,
where typical rtt is below 1 ms.
We are planning a switch to usec resolution for linux-3.15,
but in the meantime, this patch removes the 1 ms offset.
All we need is to have tp->srtt minimal value of 1 to differentiate
the case of srtt being initialized or not, not 8.
The problematic behavior was observed on a 40Gbit testbed,
where 32 concurrent netperf were reaching 12Gbps of aggregate
speed, instead of line speed.
This patch also has the effect of reporting more accurate srtt and send
rates to iproute2 ss command as in :
$ ss -i dst cca2
Netid State Recv-Q Send-Q Local Address:Port
Peer Address:Port
tcp ESTAB 0 0 10.244.129.1:56984
10.244.129.2:12865
cubic wscale:6,6 rto:200 rtt:0.25/0.25 ato:40 mss:1448 cwnd:10 send
463.4Mbps rcv_rtt:1 rcv_space:29200
tcp ESTAB 0 390960 10.244.129.1:60247
10.244.129.2:50204
cubic wscale:6,6 rto:200 rtt:0.875/0.75 mss:1448 cwnd:73 ssthresh:51
send 966.4Mbps unacked:73 retrans:0/121 rcv_space:29200
Reported-by: Vytautas Valancius <valas@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
The following are only used in one file:
tcp_connect_init
tcp_set_rto
Signed-off-by: Stephen Hemminger <stephen@networkplumber.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
While investigating performance problems on small RPC workloads,
I noticed linux TCP stack was always splitting the last TSO skb
into two parts (skbs). One being a multiple of MSS, and a small one
with the Push flag. This split is done even if TCP_NODELAY is set,
or if no small packet is in flight.
Example with request/response of 4K/4K
IP A > B: . ack 68432 win 2783 <nop,nop,timestamp 6524593 6525001>
IP A > B: . 65537:68433(2896) ack 69632 win 2783 <nop,nop,timestamp 6524593 6525001>
IP A > B: P 68433:69633(1200) ack 69632 win 2783 <nop,nop,timestamp 6524593 6525001>
IP B > A: . ack 68433 win 2768 <nop,nop,timestamp 6525001 6524593>
IP B > A: . 69632:72528(2896) ack 69633 win 2768 <nop,nop,timestamp 6525001 6524593>
IP B > A: P 72528:73728(1200) ack 69633 win 2768 <nop,nop,timestamp 6525001 6524593>
IP A > B: . ack 72528 win 2783 <nop,nop,timestamp 6524593 6525001>
IP A > B: . 69633:72529(2896) ack 73728 win 2783 <nop,nop,timestamp 6524593 6525001>
IP A > B: P 72529:73729(1200) ack 73728 win 2783 <nop,nop,timestamp 6524593 6525001>
We can avoid this split by including the Nagle tests at the right place.
Note : If some NIC had trouble sending TSO packets with a partial
last segment, we would have hit the problem in GRO/forwarding workload already.
tcp_minshall_update() is moved to tcp_output.c and is updated as we might
feed a TSO packet with a partial last segment.
This patch tremendously improves performance, as the traffic now looks
like :
IP A > B: . ack 98304 win 2783 <nop,nop,timestamp 6834277 6834685>
IP A > B: P 94209:98305(4096) ack 98304 win 2783 <nop,nop,timestamp 6834277 6834685>
IP B > A: . ack 98305 win 2768 <nop,nop,timestamp 6834686 6834277>
IP B > A: P 98304:102400(4096) ack 98305 win 2768 <nop,nop,timestamp 6834686 6834277>
IP A > B: . ack 102400 win 2783 <nop,nop,timestamp 6834279 6834686>
IP A > B: P 98305:102401(4096) ack 102400 win 2783 <nop,nop,timestamp 6834279 6834686>
IP B > A: . ack 102401 win 2768 <nop,nop,timestamp 6834687 6834279>
IP B > A: P 102400:106496(4096) ack 102401 win 2768 <nop,nop,timestamp 6834687 6834279>
IP A > B: . ack 106496 win 2783 <nop,nop,timestamp 6834280 6834687>
IP A > B: P 102401:106497(4096) ack 106496 win 2783 <nop,nop,timestamp 6834280 6834687>
IP B > A: . ack 106497 win 2768 <nop,nop,timestamp 6834688 6834280>
IP B > A: P 106496:110592(4096) ack 106497 win 2768 <nop,nop,timestamp 6834688 6834280>
Before :
lpq83:~# nstat >/dev/null;perf stat ./super_netperf 200 -t TCP_RR -H lpq84 -l 20 -- -r 4K,4K
280774
Performance counter stats for './super_netperf 200 -t TCP_RR -H lpq84 -l 20 -- -r 4K,4K':
205719.049006 task-clock # 9.278 CPUs utilized
8,449,968 context-switches # 0.041 M/sec
1,935,997 CPU-migrations # 0.009 M/sec
160,541 page-faults # 0.780 K/sec
548,478,722,290 cycles # 2.666 GHz [83.20%]
455,240,670,857 stalled-cycles-frontend # 83.00% frontend cycles idle [83.48%]
272,881,454,275 stalled-cycles-backend # 49.75% backend cycles idle [66.73%]
166,091,460,030 instructions # 0.30 insns per cycle
# 2.74 stalled cycles per insn [83.39%]
29,150,229,399 branches # 141.699 M/sec [83.30%]
1,943,814,026 branch-misses # 6.67% of all branches [83.32%]
22.173517844 seconds time elapsed
lpq83:~# nstat | egrep "IpOutRequests|IpExtOutOctets"
IpOutRequests 16851063 0.0
IpExtOutOctets 23878580777 0.0
After patch :
lpq83:~# nstat >/dev/null;perf stat ./super_netperf 200 -t TCP_RR -H lpq84 -l 20 -- -r 4K,4K
280877
Performance counter stats for './super_netperf 200 -t TCP_RR -H lpq84 -l 20 -- -r 4K,4K':
107496.071918 task-clock # 4.847 CPUs utilized
5,635,458 context-switches # 0.052 M/sec
1,374,707 CPU-migrations # 0.013 M/sec
160,920 page-faults # 0.001 M/sec
281,500,010,924 cycles # 2.619 GHz [83.28%]
228,865,069,307 stalled-cycles-frontend # 81.30% frontend cycles idle [83.38%]
142,462,742,658 stalled-cycles-backend # 50.61% backend cycles idle [66.81%]
95,227,712,566 instructions # 0.34 insns per cycle
# 2.40 stalled cycles per insn [83.43%]
16,209,868,171 branches # 150.795 M/sec [83.20%]
874,252,952 branch-misses # 5.39% of all branches [83.37%]
22.175821286 seconds time elapsed
lpq83:~# nstat | egrep "IpOutRequests|IpExtOutOctets"
IpOutRequests 11239428 0.0
IpExtOutOctets 23595191035 0.0
Indeed, the occupancy of tx skbs (IpExtOutOctets/IpOutRequests) is higher :
2099 instead of 1417, thus helping GRO to be more efficient when using FQ packet
scheduler.
Many thanks to Neal for review and ideas.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Nandita Dukkipati <nanditad@google.com>
Cc: Van Jacobson <vanj@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Tested-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Various spelling fixes in networking stack
Signed-off-by: Stephen Hemminger <stephen@networkplumber.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
Compiler doesn't know skb_shinfo(skb) pointer is usually constant.
By using a temporary variable, we help generating smaller code.
For example, tcp_init_nondata_skb() is inlined after this patch.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
snd_nxt must be updated synchronously with sk_send_head. Otherwise
tp->packets_out may be updated incorrectly, what may bring a kernel panic.
Here is a kernel panic from my host.
[ 103.043194] BUG: unable to handle kernel NULL pointer dereference at 0000000000000048
[ 103.044025] IP: [<ffffffff815aaaaf>] tcp_rearm_rto+0xcf/0x150
...
[ 146.301158] Call Trace:
[ 146.301158] [<ffffffff815ab7f0>] tcp_ack+0xcc0/0x12c0
Before this panic a tcp socket was restored. This socket had sent and
unsent data in the write queue. Sent data was restored in repair mode,
then the socket was switched from reapair mode and unsent data was
restored. After that the socket was switched back into repair mode.
In that moment we had a socket where write queue looks like this:
snd_una snd_nxt write_seq
|_________|________|
|
sk_send_head
After a second switching from repair mode the state of socket was
changed:
snd_una snd_nxt, write_seq
|_________ ________|
|
sk_send_head
This state is inconsistent, because snd_nxt and sk_send_head are not
synchronized.
Bellow you can find a call trace, how packets_out can be incremented
twice for one skb, if snd_nxt and sk_send_head are not synchronized.
In this case packets_out will be always positive, even when
sk_write_queue is empty.
tcp_write_wakeup
skb = tcp_send_head(sk);
tcp_fragment
if (!before(tp->snd_nxt, TCP_SKB_CB(buff)->end_seq))
tcp_adjust_pcount(sk, skb, diff);
tcp_event_new_data_sent
tp->packets_out += tcp_skb_pcount(skb);
I think update of snd_nxt isn't required, when a socket is switched from
repair mode. Because it's initialized in tcp_connect_init. Then when a
write queue is restored, snd_nxt is incremented in tcp_event_new_data_sent,
so it's always is in consistent state.
I have checked, that the bug is not reproduced with this patch and
all tests about restoring tcp connections work fine.
Cc: Pavel Emelyanov <xemul@parallels.com>
Cc: Eric Dumazet <edumazet@google.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: James Morris <jmorris@namei.org>
Cc: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
Cc: Patrick McHardy <kaber@trash.net>
Signed-off-by: Andrey Vagin <avagin@openvz.org>
Acked-by: Pavel Emelyanov <xemul@parallels.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
After commit c9eeec26e3 ("tcp: TSQ can use a dynamic limit"), several
users reported throughput regressions, notably on mvneta and wifi
adapters.
802.11 AMPDU requires a fair amount of queueing to be effective.
This patch partially reverts the change done in tcp_write_xmit()
so that the minimal amount is sysctl_tcp_limit_output_bytes.
It also remove the use of this sysctl while building skb stored
in write queue, as TSO autosizing does the right thing anyway.
Users with well behaving NICS and correct qdisc (like sch_fq),
can then lower the default sysctl_tcp_limit_output_bytes value from
128KB to 8KB.
This new usage of sysctl_tcp_limit_output_bytes permits each driver
authors to check how their driver performs when/if the value is set
to a minimum of 4KB.
Normally, line rate for a single TCP flow should be possible,
but some drivers rely on timers to perform TX completion and
too long TX completion delays prevent reaching full throughput.
Fixes: c9eeec26e3 ("tcp: TSQ can use a dynamic limit")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reported-by: Sujith Manoharan <sujith@msujith.org>
Reported-by: Arnaud Ebalard <arno@natisbad.org>
Tested-by: Sujith Manoharan <sujith@msujith.org>
Cc: Felix Fietkau <nbd@openwrt.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
drivers/net/usb/qmi_wwan.c
include/net/dst.h
Trivial merge conflicts, both were overlapping changes.
Signed-off-by: David S. Miller <davem@davemloft.net>
Remove the specialized code in __tcp_retransmit_skb() that tries to trim
any ACKed payload preceding a FIN before we retransmit (this was added
in 1999 in v2.2.3pre3). This trimming code was made unreachable by the
more general code added above it that uses tcp_trim_head() to trim any
ACKed payload, with or without a FIN (this was added in "[NET]: Add
segmentation offload support to TCP." in 2002 circa v2.5.33).
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
sk_can_gso() should only be used as a hint in tcp_sendmsg() to build GSO
packets in the first place. (As a performance hint)
Once we have GSO packets in write queue, we can not decide they are no
longer GSO only because flow now uses a route which doesn't handle
TSO/GSO.
Core networking stack handles the case very well for us, all we need
is keeping track of packet counts in MSS terms, regardless of
segmentation done later (in GSO or hardware)
Right now, if tcp_fragment() splits a GSO packet in two parts,
@left and @right, and route changed through a non GSO device,
both @left and @right have pcount set to 1, which is wrong,
and leads to incorrect packet_count tracking.
This problem was added in commit d5ac99a648 ("[TCP]: skb pcount with MTU
discovery")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Reported-by: Maciej Żenczykowski <maze@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
TCP stack should make sure it owns skbs before mangling them.
We had various crashes using bnx2x, and it turned out gso_size
was cleared right before bnx2x driver was populating TC descriptor
of the _previous_ packet send. TCP stack can sometime retransmit
packets that are still in Qdisc.
Of course we could make bnx2x driver more robust (using
ACCESS_ONCE(shinfo->gso_size) for example), but the bug is TCP stack.
We have identified two points where skb_unclone() was needed.
This patch adds a WARN_ON_ONCE() to warn us if we missed another
fix of this kind.
Kudos to Neal for finding the root cause of this bug. Its visible
using small MSS.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
1) We need to take a timestamp only for skb that should be cloned.
Other skbs are not in write queue and no rtt estimation is done on them.
2) the unlikely() hint is wrong for receivers (they send pure ACK)
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: MF Nowlan <fitz@cs.yale.edu>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Acked-By: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
In commit 634fb979e8 ("inet: includes a sock_common in request_sock")
I forgot that the two ports in sock_common do not have same byte order :
skc_dport is __be16 (network order), but skc_num is __u16 (host order)
So sparse complains because ir_loc_port (mapped into skc_num) is
considered as __u16 while it should be __be16
Let rename ir_loc_port to ireq->ir_num (analogy with inet->inet_num),
and perform appropriate htons/ntohs conversions.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reported-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
TCP listener refactoring, part 5 :
We want to be able to insert request sockets (SYN_RECV) into main
ehash table instead of the per listener hash table to allow RCU
lookups and remove listener lock contention.
This patch includes the needed struct sock_common in front
of struct request_sock
This means there is no more inet6_request_sock IPv6 specific
structure.
Following inet_request_sock fields were renamed as they became
macros to reference fields from struct sock_common.
Prefix ir_ was chosen to avoid name collisions.
loc_port -> ir_loc_port
loc_addr -> ir_loc_addr
rmt_addr -> ir_rmt_addr
rmt_port -> ir_rmt_port
iif -> ir_iif
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_established_options assumes opts->options is 0 before calling,
as it read modify writes it.
For the tcp_current_mss() case the opts structure is not zeroed,
so this can be done with uninitialized values.
This is ok, because ->options is not read in this path.
But it's still better to avoid the operation on the uninitialized
field. This shuts up a static code analyzer, and presumably
may help the optimizer.
Cc: netdev@vger.kernel.org
Signed-off-by: Andi Kleen <ak@linux.intel.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
When TCP Small Queues was added, we used a sysctl to limit amount of
packets queues on Qdisc/device queues for a given TCP flow.
Problem is this limit is either too big for low rates, or too small
for high rates.
Now TCP stack has rate estimation in sk->sk_pacing_rate, and TSO
auto sizing, it can better control number of packets in Qdisc/device
queues.
New limit is two packets or at least 1 to 2 ms worth of packets.
Low rates flows benefit from this patch by having even smaller
number of packets in queues, allowing for faster recovery,
better RTT estimations.
High rates flows benefit from this patch by allowing more than 2 packets
in flight as we had reports this was a limiting factor to reach line
rate. [ In particular if TX completion is delayed because of coalescing
parameters ]
Example for a single flow on 10Gbp link controlled by FQ/pacing
14 packets in flight instead of 2
$ tc -s -d qd
qdisc fq 8001: dev eth0 root refcnt 32 limit 10000p flow_limit 100p
buckets 1024 quantum 3028 initial_quantum 15140
Sent 1168459366606 bytes 771822841 pkt (dropped 0, overlimits 0
requeues 6822476)
rate 9346Mbit 771713pps backlog 953820b 14p requeues 6822476
2047 flow, 2046 inactive, 1 throttled, delay 15673 ns
2372 gc, 0 highprio, 0 retrans, 9739249 throttled, 0 flows_plimit
Note that sk_pacing_rate is currently set to twice the actual rate, but
this might be refined in the future when a flow is in congestion
avoidance.
Additional change : skb->destructor should be set to tcp_wfree().
A future patch (for linux 3.13+) might remove tcp_limit_output_bytes
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Wei Liu <wei.liu2@citrix.com>
Cc: Cong Wang <xiyou.wangcong@gmail.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
drivers/net/ethernet/stmicro/stmmac/stmmac_platform.c
net/bridge/br_multicast.c
net/ipv6/sit.c
The conflicts were minor:
1) sit.c changes overlap with change to ip_tunnel_xmit() signature.
2) br_multicast.c had an overlap between computing max_delay using
msecs_to_jiffies and turning MLDV2_MRC() into an inline function
with a name using lowercase instead of uppercase letters.
3) stmmac had two overlapping changes, one which conditionally allocated
and hooked up a dma_cfg based upon the presence of the pbl OF property,
and another one handling store-and-forward DMA made. The latter of
which should not go into the new of_find_property() basic block.
Signed-off-by: David S. Miller <davem@davemloft.net>
In commit 90ba9b19 (tcp: tcp_make_synack() can use alloc_skb()), Eric changed
the call to sock_wmalloc in tcp_make_synack to alloc_skb. In doing so,
the netfilter owner match lost its ability to block the SYNACK packet on
outbound listening sockets. Revert the change, restoring the owner match
functionality.
This closes netfilter bugzilla #847.
Signed-off-by: Phil Oester <kernel@linuxace.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
After hearing many people over past years complaining against TSO being
bursty or even buggy, we are proud to present automatic sizing of TSO
packets.
One part of the problem is that tcp_tso_should_defer() uses an heuristic
relying on upcoming ACKS instead of a timer, but more generally, having
big TSO packets makes little sense for low rates, as it tends to create
micro bursts on the network, and general consensus is to reduce the
buffering amount.
This patch introduces a per socket sk_pacing_rate, that approximates
the current sending rate, and allows us to size the TSO packets so
that we try to send one packet every ms.
This field could be set by other transports.
Patch has no impact for high speed flows, where having large TSO packets
makes sense to reach line rate.
For other flows, this helps better packet scheduling and ACK clocking.
This patch increases performance of TCP flows in lossy environments.
A new sysctl (tcp_min_tso_segs) is added, to specify the
minimal size of a TSO packet (default being 2).
A follow-up patch will provide a new packet scheduler (FQ), using
sk_pacing_rate as an input to perform optional per flow pacing.
This explains why we chose to set sk_pacing_rate to twice the current
rate, allowing 'slow start' ramp up.
sk_pacing_rate = 2 * cwnd * mss / srtt
v2: Neal Cardwell reported a suspect deferring of last two segments on
initial write of 10 MSS, I had to change tcp_tso_should_defer() to take
into account tp->xmit_size_goal_segs
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Van Jacobson <vanj@google.com>
Cc: Tom Herbert <therbert@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
u32 rcv_tstamp; /* timestamp of last received ACK */
Its value used in tcp_retransmit_timer, which closes socket
if the last ack was received more then TCP_RTO_MAX ago.
Currently rcv_tstamp is initialized to zero and if tcp_retransmit_timer
is called before receiving a first ack, the connection is closed.
This patch initializes rcv_tstamp to a timestamp, when a socket was
restored.
Cc: Pavel Emelyanov <xemul@parallels.com>
Cc: Eric Dumazet <eric.dumazet@gmail.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: James Morris <jmorris@namei.org>
Cc: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
Cc: Patrick McHardy <kaber@trash.net>
Reported-by: Cyrill Gorcunov <gorcunov@openvz.org>
Signed-off-by: Andrey Vagin <avagin@openvz.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
Idea of this patch is to add optional limitation of number of
unsent bytes in TCP sockets, to reduce usage of kernel memory.
TCP receiver might announce a big window, and TCP sender autotuning
might allow a large amount of bytes in write queue, but this has little
performance impact if a large part of this buffering is wasted :
Write queue needs to be large only to deal with large BDP, not
necessarily to cope with scheduling delays (incoming ACKS make room
for the application to queue more bytes)
For most workloads, using a value of 128 KB or less is OK to give
applications enough time to react to POLLOUT events in time
(or being awaken in a blocking sendmsg())
This patch adds two ways to set the limit :
1) Per socket option TCP_NOTSENT_LOWAT
2) A sysctl (/proc/sys/net/ipv4/tcp_notsent_lowat) for sockets
not using TCP_NOTSENT_LOWAT socket option (or setting a zero value)
Default value being UINT_MAX (0xFFFFFFFF), meaning this has no effect.
This changes poll()/select()/epoll() to report POLLOUT
only if number of unsent bytes is below tp->nosent_lowat
Note this might increase number of sendmsg()/sendfile() calls
when using non blocking sockets,
and increase number of context switches for blocking sockets.
Note this is not related to SO_SNDLOWAT (as SO_SNDLOWAT is
defined as :
Specify the minimum number of bytes in the buffer until
the socket layer will pass the data to the protocol)
Tested:
netperf sessions, and watching /proc/net/protocols "memory" column for TCP
With 200 concurrent netperf -t TCP_STREAM sessions, amount of kernel memory
used by TCP buffers shrinks by ~55 % (20567 pages instead of 45458)
lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols
TCPv6 1880 2 45458 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y
TCP 1696 508 45458 no 208 yes kernel y y y y y y y y y y y y y n y y y y y
lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols
TCPv6 1880 2 20567 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y
TCP 1696 508 20567 no 208 yes kernel y y y y y y y y y y y y y n y y y y y
Using 128KB has no bad effect on the throughput or cpu usage
of a single flow, although there is an increase of context switches.
A bonus is that we hold socket lock for a shorter amount
of time and should improve latencies of ACK processing.
lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3
OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf.
Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service
Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand
Size Size Size (sec) Util Util Util Util Demand Demand Units
Final Final % Method % Method
1651584 6291456 16384 20.00 17447.90 10^6bits/s 3.13 S -1.00 U 0.353 -1.000 usec/KB
Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3':
412,514 context-switches
200.034645535 seconds time elapsed
lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3
OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf.
Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service
Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand
Size Size Size (sec) Util Util Util Util Demand Demand Units
Final Final % Method % Method
1593240 6291456 16384 20.00 17321.16 10^6bits/s 3.35 S -1.00 U 0.381 -1.000 usec/KB
Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3':
2,675,818 context-switches
200.029651391 seconds time elapsed
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-By: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Change snmp RETRANSFAILS stat to include timeout retransmit failures
in addition to other loss recoveries.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
In previous discussions, I tried to find some reasonable heuristics
for delayed ACK, however this seems not possible, according to Eric:
"ACKS might also be delayed because of bidirectional
traffic, and is more controlled by the application
response time. TCP stack can not easily estimate it."
"ACK can be incredibly useful to recover from losses in
a short time.
The vast majority of TCP sessions are small lived, and we
send one ACK per received segment anyway at beginning or
retransmits to let the sender smoothly increase its cwnd,
so an auto-tuning facility wont help them that much."
and according to David:
"ACKs are the only information we have to detect loss.
And, for the same reasons that TCP VEGAS is fundamentally
broken, we cannot measure the pipe or some other
receiver-side-visible piece of information to determine
when it's "safe" to stretch ACK.
And even if it's "safe", we should not do it so that losses are
accurately detected and we don't spuriously retransmit.
The only way to know when the bandwidth increases is to
"test" it, by sending more and more packets until drops happen.
That's why all successful congestion control algorithms must
operate on explicited tested pieces of information.
Similarly, it's not really possible to universally know if
it's safe to stretch ACK or not."
It still makes sense to enable or disable quick ack mode like
what TCP_QUICK_ACK does.
Similar to TCP_QUICK_ACK option, but for people who can't
modify the source code and still wants to control
TCP delayed ACK behavior. As David suggested, this should belong
to per-path scope, since different pathes may want different
behaviors.
Cc: Eric Dumazet <eric.dumazet@gmail.com>
Cc: Rick Jones <rick.jones2@hp.com>
Cc: Stephen Hemminger <stephen@networkplumber.org>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Thomas Graf <tgraf@suug.ch>
CC: David Laight <David.Laight@ACULAB.COM>
Signed-off-by: Cong Wang <amwang@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Linux sends new unset data during disorder and recovery state if all
(suspected) lost packets have been retransmitted ( RFC5681, section
3.2 step 1 & 2, RFC3517 section 4, NexSeg() Rule 2). One requirement
is to keep the receive window about twice the estimated sender's
congestion window (tcp_rcv_space_adjust()), assuming the fast
retransmits repair the losses in the next round trip.
But currently it's not the case on the first round trip in either
normal or Fast Open connection, beucase the initial receive window
is identical to (expected) sender's initial congestion window. The
fix is to double it.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
commit 3853b5841c ("xps: Improvements in TX queue selection")
introduced ooo_okay flag, but the condition to set it is slightly wrong.
In our traces, we have seen ACK packets being received out of order,
and RST packets sent in response.
We should test if we have any packets still in host queue.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Tom Herbert <therbert@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Add MIB counters for checksum errors in IP layer,
and TCP/UDP/ICMP layers, to help diagnose problems.
$ nstat -a | grep Csum
IcmpInCsumErrors 72 0.0
TcpInCsumErrors 382 0.0
UdpInCsumErrors 463221 0.0
Icmp6InCsumErrors 75 0.0
Udp6InCsumErrors 173442 0.0
IpExtInCsumErrors 10884 0.0
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
drivers/net/ethernet/emulex/benet/be_main.c
drivers/net/ethernet/intel/igb/igb_main.c
drivers/net/wireless/brcm80211/brcmsmac/mac80211_if.c
include/net/scm.h
net/batman-adv/routing.c
net/ipv4/tcp_input.c
The e{uid,gid} --> {uid,gid} credentials fix conflicted with the
cleanup in net-next to now pass cred structs around.
The be2net driver had a bug fix in 'net' that overlapped with the VLAN
interface changes by Patrick McHardy in net-next.
An IGB conflict existed because in 'net' the build_skb() support was
reverted, and in 'net-next' there was a comment style fix within that
code.
Several batman-adv conflicts were resolved by making sure that all
calls to batadv_is_my_mac() are changed to have a new bat_priv first
argument.
Eric Dumazet's TS ECR fix in TCP in 'net' conflicted with the F-RTO
rewrite in 'net-next', mostly overlapping changes.
Thanks to Stephen Rothwell and Antonio Quartulli for help with several
of these merge resolutions.
Signed-off-by: David S. Miller <davem@davemloft.net>
Host queues (Qdisc + NIC) can hold packets so long that TCP can
eventually retransmit a packet before the first transmit even left
the host.
Its not clear right now if we could avoid this in the first place :
- We could arm RTO timer not at the time we enqueue packets, but
at the time we TX complete them (tcp_wfree())
- Cancel the sending of the new copy of the packet if prior one
is still in queue.
This patch adds instrumentation so that we can at least see how
often this problem happens.
TCPSpuriousRtxHostQueues SNMP counter is incremented every time
we detect the fast clone is not yet freed in tcp_transmit_skb()
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Tom Herbert <therbert@google.com>
Cc: Willem de Bruijn <willemb@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
I noticed that TSQ (TCP Small queues) was less effective when TSO is
turned off, and GSO is on. If BQL is not enabled, TSQ has then no
effect.
It turns out the GSO engine frees the original gso_skb at the time the
fragments are generated and queued to the NIC.
We should instead call the tcp_wfree() destructor for the last fragment,
to keep the flow control as intended in TSQ. This effectively limits
the number of queued packets on qdisc + NIC layers.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Tom Herbert <therbert@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Nandita Dukkipati <nanditad@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
If a TCP retransmission gets partially ACKed and collapsed multiple
times it is possible for the headroom to grow beyond 64K which will
overflow the 16bit skb->csum_start which is based on the start of
the headroom. It has been observed rarely in the wild with IPoIB due
to the 64K MTU.
Verify if the acking and collapsing resulted in a headroom exceeding
what csum_start can cover and reallocate the headroom if so.
A big thank you to Jim Foraker <foraker1@llnl.gov> and the team at
LLNL for helping out with the investigation and testing.
Reported-by: Jim Foraker <foraker1@llnl.gov>
Signed-off-by: Thomas Graf <tgraf@suug.ch>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Commit 90ba9b1986 (tcp: tcp_make_synack() can use alloc_skb())
broke certain SELinux/NetLabel configurations by no longer correctly
assigning the sock to the outgoing SYNACK packet.
Cost of atomic operations on the LISTEN socket is quite big,
and we would like it to happen only if really needed.
This patch introduces a new security_ops->skb_owned_by() method,
that is a void operation unless selinux is active.
Reported-by: Miroslav Vadkerti <mvadkert@redhat.com>
Diagnosed-by: Paul Moore <pmoore@redhat.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: linux-security-module@vger.kernel.org
Acked-by: James Morris <james.l.morris@oracle.com>
Tested-by: Paul Moore <pmoore@redhat.com>
Acked-by: Paul Moore <pmoore@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Pull to get the thermal netlink multicast group name fix, otherwise
the assertion added in net-next to netlink to detect that kind of bug
makes systems unbootable for some folks.
Signed-off-by: David S. Miller <davem@davemloft.net>
A long standing problem with TSO is the fact that tcp_tso_should_defer()
rearms the deferred timer, while it should not.
Current code leads to following bad bursty behavior :
20:11:24.484333 IP A > B: . 297161:316921(19760) ack 1 win 119
20:11:24.484337 IP B > A: . ack 263721 win 1117
20:11:24.485086 IP B > A: . ack 265241 win 1117
20:11:24.485925 IP B > A: . ack 266761 win 1117
20:11:24.486759 IP B > A: . ack 268281 win 1117
20:11:24.487594 IP B > A: . ack 269801 win 1117
20:11:24.488430 IP B > A: . ack 271321 win 1117
20:11:24.489267 IP B > A: . ack 272841 win 1117
20:11:24.490104 IP B > A: . ack 274361 win 1117
20:11:24.490939 IP B > A: . ack 275881 win 1117
20:11:24.491775 IP B > A: . ack 277401 win 1117
20:11:24.491784 IP A > B: . 316921:332881(15960) ack 1 win 119
20:11:24.492620 IP B > A: . ack 278921 win 1117
20:11:24.493448 IP B > A: . ack 280441 win 1117
20:11:24.494286 IP B > A: . ack 281961 win 1117
20:11:24.495122 IP B > A: . ack 283481 win 1117
20:11:24.495958 IP B > A: . ack 285001 win 1117
20:11:24.496791 IP B > A: . ack 286521 win 1117
20:11:24.497628 IP B > A: . ack 288041 win 1117
20:11:24.498459 IP B > A: . ack 289561 win 1117
20:11:24.499296 IP B > A: . ack 291081 win 1117
20:11:24.500133 IP B > A: . ack 292601 win 1117
20:11:24.500970 IP B > A: . ack 294121 win 1117
20:11:24.501388 IP B > A: . ack 295641 win 1117
20:11:24.501398 IP A > B: . 332881:351881(19000) ack 1 win 119
While the expected behavior is more like :
20:19:49.259620 IP A > B: . 197601:202161(4560) ack 1 win 119
20:19:49.260446 IP B > A: . ack 154281 win 1212
20:19:49.261282 IP B > A: . ack 155801 win 1212
20:19:49.262125 IP B > A: . ack 157321 win 1212
20:19:49.262136 IP A > B: . 202161:206721(4560) ack 1 win 119
20:19:49.262958 IP B > A: . ack 158841 win 1212
20:19:49.263795 IP B > A: . ack 160361 win 1212
20:19:49.264628 IP B > A: . ack 161881 win 1212
20:19:49.264637 IP A > B: . 206721:211281(4560) ack 1 win 119
20:19:49.265465 IP B > A: . ack 163401 win 1212
20:19:49.265886 IP B > A: . ack 164921 win 1212
20:19:49.266722 IP B > A: . ack 166441 win 1212
20:19:49.266732 IP A > B: . 211281:215841(4560) ack 1 win 119
20:19:49.267559 IP B > A: . ack 167961 win 1212
20:19:49.268394 IP B > A: . ack 169481 win 1212
20:19:49.269232 IP B > A: . ack 171001 win 1212
20:19:49.269241 IP A > B: . 215841:221161(5320) ack 1 win 119
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Van Jacobson <vanj@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
The patch series refactor the F-RTO feature (RFC4138/5682).
This is to simplify the loss recovery processing. Existing F-RTO
was developed during the experimental stage (RFC4138) and has
many experimental features. It takes a separate code path from
the traditional timeout processing by overloading CA_Disorder
instead of using CA_Loss state. This complicates CA_Disorder state
handling because it's also used for handling dubious ACKs and undos.
While the algorithm in the RFC does not change the congestion control,
the implementation intercepts congestion control in various places
(e.g., frto_cwnd in tcp_ack()).
The new code implements newer F-RTO RFC5682 using CA_Loss processing
path. F-RTO becomes a small extension in the timeout processing
and interfaces with congestion control and Eifel undo modules.
It lets congestion control (module) determines how many to send
independently. F-RTO only chooses what to send in order to detect
spurious retranmission. If timeout is found spurious it invokes
existing Eifel undo algorithms like DSACK or TCP timestamp based
detection.
The first patch removes all F-RTO code except the sysctl_tcp_frto is
left for the new implementation. Since CA_EVENT_FRTO is removed, TCP
westwood now computes ssthresh on regular timeout CA_EVENT_LOSS event.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
TCPCT uses option-number 253, reserved for experimental use and should
not be used in production environments.
Further, TCPCT does not fully implement RFC 6013.
As a nice side-effect, removing TCPCT increases TCP's performance for
very short flows:
Doing an apache-benchmark with -c 100 -n 100000, sending HTTP-requests
for files of 1KB size.
before this patch:
average (among 7 runs) of 20845.5 Requests/Second
after:
average (among 7 runs) of 21403.6 Requests/Second
Signed-off-by: Christoph Paasch <christoph.paasch@uclouvain.be>
Signed-off-by: David S. Miller <davem@davemloft.net>
Chrome OS team reported a crash on a Pixel ChromeBook in TCP stack :
https://code.google.com/p/chromium/issues/detail?id=182056
commit a21d45726a (tcp: avoid order-1 allocations on wifi and tx
path) did a poor choice adding an 'avail_size' field to skb, while
what we really needed was a 'reserved_tailroom' one.
It would have avoided commit 22b4a4f22d (tcp: fix retransmit of
partially acked frames) and this commit.
Crash occurs because skb_split() is not aware of the 'avail_size'
management (and should not be aware)
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reported-by: Mukesh Agrawal <quiche@chromium.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
This is the second of the TLP patch series; it augments the basic TLP
algorithm with a loss detection scheme.
This patch implements a mechanism for loss detection when a Tail
loss probe retransmission plugs a hole thereby masking packet loss
from the sender. The loss detection algorithm relies on counting
TLP dupacks as outlined in Sec. 3 of:
http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01
The basic idea is: Sender keeps track of TLP "episode" upon
retransmission of a TLP packet. An episode ends when the sender receives
an ACK above the SND.NXT (tracked by tlp_high_seq) at the time of the
episode. We want to make sure that before the episode ends the sender
receives a "TLP dupack", indicating that the TLP retransmission was
unnecessary, so there was no loss/hole that needed plugging. If the
sender gets no TLP dupack before the end of the episode, then it reduces
ssthresh and the congestion window, because the TLP packet arriving at
the receiver probably plugged a hole.
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch series implement the Tail loss probe (TLP) algorithm described
in http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01. The
first patch implements the basic algorithm.
TLP's goal is to reduce tail latency of short transactions. It achieves
this by converting retransmission timeouts (RTOs) occuring due
to tail losses (losses at end of transactions) into fast recovery.
TLP transmits one packet in two round-trips when a connection is in
Open state and isn't receiving any ACKs. The transmitted packet, aka
loss probe, can be either new or a retransmission. When there is tail
loss, the ACK from a loss probe triggers FACK/early-retransmit based
fast recovery, thus avoiding a costly RTO. In the absence of loss,
there is no change in the connection state.
PTO stands for probe timeout. It is a timer event indicating
that an ACK is overdue and triggers a loss probe packet. The PTO value
is set to max(2*SRTT, 10ms) and is adjusted to account for delayed
ACK timer when there is only one oustanding packet.
TLP Algorithm
On transmission of new data in Open state:
-> packets_out > 1: schedule PTO in max(2*SRTT, 10ms).
-> packets_out == 1: schedule PTO in max(2*RTT, 1.5*RTT + 200ms)
-> PTO = min(PTO, RTO)
Conditions for scheduling PTO:
-> Connection is in Open state.
-> Connection is either cwnd limited or no new data to send.
-> Number of probes per tail loss episode is limited to one.
-> Connection is SACK enabled.
When PTO fires:
new_segment_exists:
-> transmit new segment.
-> packets_out++. cwnd remains same.
no_new_packet:
-> retransmit the last segment.
Its ACK triggers FACK or early retransmit based recovery.
ACK path:
-> rearm RTO at start of ACK processing.
-> reschedule PTO if need be.
In addition, the patch includes a small variation to the Early Retransmit
(ER) algorithm, such that ER and TLP together can in principle recover any
N-degree of tail loss through fast recovery. TLP is controlled by the same
sysctl as ER, tcp_early_retrans sysctl.
tcp_early_retrans==0; disables TLP and ER.
==1; enables RFC5827 ER.
==2; delayed ER.
==3; TLP and delayed ER. [DEFAULT]
==4; TLP only.
The TLP patch series have been extensively tested on Google Web servers.
It is most effective for short Web trasactions, where it reduced RTOs by 15%
and improved HTTP response time (average by 6%, 99th percentile by 10%).
The transmitted probes account for <0.5% of the overall transmissions.
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
In fast open the sender unncessarily reduces the space available
for data in SYN by 12 bytes. This is because in the sender
incorrectly reserves space for TS option twice in tcp_send_syn_data():
tcp_mtu_to_mss() already accounts for TS option space. But it further
reserves MAX_TCP_OPTION_SPACE when computing the payload space.
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 function will be used in next GRE_GSO patch. This patch does
not change any functionality.
Signed-off-by: Pravin B Shelar <pshelar@nicira.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Patch cef401de7b (net: fix possible wrong checksum
generation) fixed wrong checksum calculation but it broke TSO by
defining new GSO type but not a netdev feature for that type.
net_gso_ok() would not allow hardware checksum/segmentation
offload of such packets without the feature.
Following patch fixes TSO and wrong checksum. This patch uses
same logic that Eric Dumazet used. Patch introduces new flag
SKBTX_SHARED_FRAG if at least one frag can be modified by
the user. but SKBTX_SHARED_FRAG flag is kept in skb shared
info tx_flags rather than gso_type.
tx_flags is better compared to gso_type since we can have skb with
shared frag without gso packet. It does not link SHARED_FRAG to
GSO, So there is no need to define netdev feature for this.
Signed-off-by: Pravin B Shelar <pshelar@nicira.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
A socket timestamp is a sum of the global tcp_time_stamp and
a per-socket offset.
A socket offset is added in places where externally visible
tcp timestamp option is parsed/initialized.
Connections in the SYN_RECV state are not supported, global
tcp_time_stamp is used for them, because repair mode doesn't support
this state. In a future it can be implemented by the similar way
as for TIME_WAIT sockets.
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: James Morris <jmorris@namei.org>
Cc: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
Cc: Patrick McHardy <kaber@trash.net>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Pavel Emelyanov <xemul@parallels.com>
Signed-off-by: Andrey Vagin <avagin@openvz.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
Pravin Shelar mentioned that GSO could potentially generate
wrong TX checksum if skb has fragments that are overwritten
by the user between the checksum computation and transmit.
He suggested to linearize skbs but this extra copy can be
avoided for normal tcp skbs cooked by tcp_sendmsg().
This patch introduces a new SKB_GSO_SHARED_FRAG flag, set
in skb_shinfo(skb)->gso_type if at least one frag can be
modified by the user.
Typical sources of such possible overwrites are {vm}splice(),
sendfile(), and macvtap/tun/virtio_net drivers.
Tested:
$ netperf -H 7.7.8.84
MIGRATED TCP STREAM TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to
7.7.8.84 () port 0 AF_INET
Recv Send Send
Socket Socket Message Elapsed
Size Size Size Time Throughput
bytes bytes bytes secs. 10^6bits/sec
87380 16384 16384 10.00 3959.52
$ netperf -H 7.7.8.84 -t TCP_SENDFILE
TCP SENDFILE TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.8.84 ()
port 0 AF_INET
Recv Send Send
Socket Socket Message Elapsed
Size Size Size Time Throughput
bytes bytes bytes secs. 10^6bits/sec
87380 16384 16384 10.00 3216.80
Performance of the SENDFILE is impacted by the extra allocation and
copy, and because we use order-0 pages, while the TCP_STREAM uses
bigger pages.
Reported-by: Pravin Shelar <pshelar@nicira.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
As per suggestion from Eric Dumazet this patch makes tcp_ecn sysctl
namespace aware. The reason behind this patch is to ease the testing
of ecn problems on the internet and allows applications to tune their
own use of ecn.
Cc: Eric Dumazet <eric.dumazet@gmail.com>
Cc: David Miller <davem@davemloft.net>
Cc: Stephen Hemminger <shemminger@vyatta.com>
Signed-off-by: Hannes Frederic Sowa <hannes@stressinduktion.org>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Pull networking changes from David Miller:
1) Allow to dump, monitor, and change the bridge multicast database
using netlink. From Cong Wang.
2) RFC 5961 TCP blind data injection attack mitigation, from Eric
Dumazet.
3) Networking user namespace support from Eric W. Biederman.
4) tuntap/virtio-net multiqueue support by Jason Wang.
5) Support for checksum offload of encapsulated packets (basically,
tunneled traffic can still be checksummed by HW). From Joseph
Gasparakis.
6) Allow BPF filter access to VLAN tags, from Eric Dumazet and
Daniel Borkmann.
7) Bridge port parameters over netlink and BPDU blocking support
from Stephen Hemminger.
8) Improve data access patterns during inet socket demux by rearranging
socket layout, from Eric Dumazet.
9) TIPC protocol updates and cleanups from Ying Xue, Paul Gortmaker, and
Jon Maloy.
10) Update TCP socket hash sizing to be more in line with current day
realities. The existing heurstics were choosen a decade ago.
From Eric Dumazet.
11) Fix races, queue bloat, and excessive wakeups in ATM and
associated drivers, from Krzysztof Mazur and David Woodhouse.
12) Support DOVE (Distributed Overlay Virtual Ethernet) extensions
in VXLAN driver, from David Stevens.
13) Add "oops_only" mode to netconsole, from Amerigo Wang.
14) Support set and query of VEB/VEPA bridge mode via PF_BRIDGE, also
allow DCB netlink to work on namespaces other than the initial
namespace. From John Fastabend.
15) Support PTP in the Tigon3 driver, from Matt Carlson.
16) tun/vhost zero copy fixes and improvements, plus turn it on
by default, from Michael S. Tsirkin.
17) Support per-association statistics in SCTP, from Michele
Baldessari.
And many, many, driver updates, cleanups, and improvements. Too
numerous to mention individually.
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/net-next: (1722 commits)
net/mlx4_en: Add support for destination MAC in steering rules
net/mlx4_en: Use generic etherdevice.h functions.
net: ethtool: Add destination MAC address to flow steering API
bridge: add support of adding and deleting mdb entries
bridge: notify mdb changes via netlink
ndisc: Unexport ndisc_{build,send}_skb().
uapi: add missing netconf.h to export list
pkt_sched: avoid requeues if possible
solos-pci: fix double-free of TX skb in DMA mode
bnx2: Fix accidental reversions.
bna: Driver Version Updated to 3.1.2.1
bna: Firmware update
bna: Add RX State
bna: Rx Page Based Allocation
bna: TX Intr Coalescing Fix
bna: Tx and Rx Optimizations
bna: Code Cleanup and Enhancements
ath9k: check pdata variable before dereferencing it
ath5k: RX timestamp is reported at end of frame
ath9k_htc: RX timestamp is reported at end of frame
...
If SYN-ACK partially acks SYN-data, the client retransmits the
remaining data by tcp_retransmit_skb(). This increments lost recovery
state variables like tp->retrans_out in Open state. If loss recovery
happens before the retransmission is acked, it triggers the WARN_ON
check in tcp_fastretrans_alert(). For example: the client sends
SYN-data, gets SYN-ACK acking only ISN, retransmits data, sends
another 4 data packets and get 3 dupacks.
Since the retransmission is not caused by network drop it should not
update the recovery state variables. Further the server may return a
smaller MSS than the cached MSS used for SYN-data, so the retranmission
needs a loop. Otherwise some data will not be retransmitted until timeout
or other loss recovery events.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This is work the same as for ipv4.
All other hacks about tcp repair are in common code for ipv4 and ipv6,
so this patch is enough for repairing ipv6 connections.
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: James Morris <jmorris@namei.org>
Cc: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
Cc: Patrick McHardy <kaber@trash.net>
Cc: Pavel Emelyanov <xemul@parallels.com>
Signed-off-by: Andrey Vagin <avagin@openvz.org>
Acked-by: Pavel Emelyanov <xemul@parallels.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Currently if a socket was repaired with a few packet in a write queue,
a kernel bug may be triggered:
kernel BUG at net/ipv4/tcp_output.c:2330!
RIP: 0010:[<ffffffff8155784f>] tcp_retransmit_skb+0x5ff/0x610
According to the initial realization v3.4-rc2-963-gc0e88ff,
all skb-s should look like already posted. This patch fixes code
according with this sentence.
Here are three points, which were not done in the initial patch:
1. A tcp send head should not be changed
2. Initialize TSO state of a skb
3. Reset the retransmission time
This patch moves logic from tcp_sendmsg to tcp_write_xmit. A packet
passes the ussual way, but isn't sent to network. This patch solves
all described problems and handles tcp_sendpages.
Cc: Pavel Emelyanov <xemul@parallels.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: James Morris <jmorris@namei.org>
Cc: Hideaki YOSHIFUJI <yoshfuji@linux-ipv6.org>
Cc: Patrick McHardy <kaber@trash.net>
Signed-off-by: Andrey Vagin <avagin@openvz.org>
Acked-by: Pavel Emelyanov <xemul@parallels.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Use proportional rate reduction (PRR) algorithm to reduce cwnd in CWR state,
in addition to Recovery state. Retire the current rate-halving in CWR.
When losses are detected via ACKs in CWR state, the sender enters Recovery
state but the cwnd reduction continues and does not restart.
Rename and refactor cwnd reduction functions since both CWR and Recovery
use the same algorithm:
tcp_init_cwnd_reduction() is new and initiates reduction state variables.
tcp_cwnd_reduction() is previously tcp_update_cwnd_in_recovery().
tcp_ends_cwnd_reduction() is previously tcp_complete_cwr().
The rate halving functions and logic such as tcp_cwnd_down(), tcp_min_cwnd(),
and the cwnd moderation inside tcp_enter_cwr() are removed. The unused
parameter, flag, in tcp_cwnd_reduction() is also removed.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch builds on top of the previous patch to add the support
for TFO listeners. This includes -
1. allocating, properly initializing, and managing the per listener
fastopen_queue structure when TFO is enabled
2. changes to the inet_csk_accept code to support TFO. E.g., the
request_sock can no longer be freed upon accept(), not until 3WHS
finishes
3. allowing a TCP_SYN_RECV socket to properly poll() and sendmsg()
if it's a TFO socket
4. properly closing a TFO listener, and a TFO socket before 3WHS
finishes
5. supporting TCP_FASTOPEN socket option
6. modifying tcp_check_req() to use to check a TFO socket as well
as request_sock
7. supporting TCP's TFO cookie option
8. adding a new SYN-ACK retransmit handler to use the timer directly
off the TFO socket rather than the listener socket. Note that TFO
server side will not retransmit anything other than SYN-ACK until
the 3WHS is completed.
The patch also contains an important function
"reqsk_fastopen_remove()" to manage the somewhat complex relation
between a listener, its request_sock, and the corresponding child
socket. See the comment above the function for the detail.
Signed-off-by: H.K. Jerry Chu <hkchu@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Tom Herbert <therbert@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Fix sparse warning:
* symbol 'tcp_wfree' was not declared. Should it be static?
Signed-off-by: Silviu-Mihai Popescu <silviupopescu1990@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Cache the device gso_max_segs in sock::sk_gso_max_segs and use it to
limit the size of TSO skbs. This avoids the need to fall back to
software GSO for local TCP senders.
Signed-off-by: Ben Hutchings <bhutchings@solarflare.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>
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>
Modern TCP stack highly depends on tcp_write_timer() having a small
latency, but current implementation doesn't exactly meet the
expectations.
When a timer fires but finds the socket is owned by the user, it rearms
itself for an additional delay hoping next run will be more
successful.
tcp_write_timer() for example uses a 50ms delay for next try, and it
defeats many attempts to get predictable TCP behavior in term of
latencies.
Use the recently introduced tcp_release_cb(), so that the user owning
the socket will call various handlers right before socket release.
This will permit us to post a followup patch to address the
tcp_tso_should_defer() syndrome (some deferred packets have to wait
RTO timer to be transmitted, while cwnd should allow us to send them
sooner)
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Tom Herbert <therbert@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Nandita Dukkipati <nanditad@google.com>
Cc: H.K. Jerry Chu <hkchu@google.com>
Cc: John Heffner <johnwheffner@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
In trusted networks, e.g., intranet, data-center, the client does not
need to use Fast Open cookie to mitigate DoS attacks. In cookie-less
mode, sendmsg() with MSG_FASTOPEN flag will send SYN-data regardless
of cookie availability.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
On paths with firewalls dropping SYN with data or experimental TCP options,
Fast Open connections will have experience SYN timeout and bad performance.
The solution is to track such incidents in the cookie cache and disables
Fast Open temporarily.
Since only the original SYN includes data and/or Fast Open option, the
SYN-ACK has some tell-tale sign (tcp_rcv_fastopen_synack()) to detect
such drops. If a path has recurring Fast Open SYN drops, Fast Open is
disabled for 2^(recurring_losses) minutes starting from four minutes up to
roughly one and half day. sendmsg with MSG_FASTOPEN flag will succeed but
it behaves as connect() then write().
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 patch implements sending SYN-data in tcp_connect(). The data is
from tcp_sendmsg() with flag MSG_FASTOPEN (implemented in a later patch).
The length of the cookie in tcp_fastopen_req, init'd to 0, controls the
type of the SYN. If the cookie is not cached (len==0), the host sends
data-less SYN with Fast Open cookie request option to solicit a cookie
from the remote. If cookie is not available (len > 0), the host sends
a SYN-data with Fast Open cookie option. If cookie length is negative,
the SYN will not include any Fast Open option (for fall back operations).
To deal with middleboxes that may drop SYN with data or experimental TCP
option, the SYN-data is only sent once. SYN retransmits do not include
data or Fast Open options. The connection will fall back to regular TCP
handshake.
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 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>
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: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Matt Mathis <mattmathis@google.com>
Cc: Mahesh Bandewar <maheshb@google.com>
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>
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>
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: Eric Dumazet <edumazet@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>
Use the current debugging style and enable dynamic_debug.
Signed-off-by: Joe Perches <joe@perches.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>
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: Yuchung Cheng <ycheng@google.com>
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>
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: Pavel Emelyanov <xemul@parallels.com>
Signed-off-by: David 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: Pavel Emelyanov <xemul@parallels.com>
Signed-off-by: David 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: Pavel Emelyanov <xemul@parallels.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Alexander Beregalov reported skb_over_panic errors and provided stack
trace.
I occurs commit a21d45726a (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: Alexander Beregalov <a.beregalov@gmail.com>
Cc: Marc MERLIN <marc@merlins.org>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Use of "unsigned int" is preferred to bare "unsigned" in net tree.
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David 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: Marc MERLIN <marc@merlins.org>
Tested-by: Marc MERLIN <marc@merlins.org>
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David 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: Neal Cardwell <ncardwell@google.com>
Acked-by: Nandita Dukkipati <nanditad@google.com>
Acked-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
It might be useful to get a counter of failed tcp_retransmit_skb()
calls.
Reported-by: Satoru Moriya <satoru.moriya@hds.com>
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.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>
commit f07d960df3 (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: Vijay Subramanian <subramanian.vijay@gmail.com>
Signed-off-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David 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: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Since 2005 (c1b4a7e695)
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
(dd9e0dda66) 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: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David 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: Feng King <kinwin2008@gmail.com>
Signed-off-by: David 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: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David 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: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David 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: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: David 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.pdfhttp://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: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: David 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: David 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: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
Eric Dumazet