The congestion state and cwnd can be updated in the wrong order.
For example, upon receiving a dubious ACK, we incorrectly raise
the cwnd first (tcp_may_raise_cwnd()/tcp_cong_avoid()) because
the state is still Open, then enter recovery state to reduce cwnd.
For another example, if the ACK indicates spurious timeout or
retransmits, we first revert the cwnd reduction and congestion
state back to Open state. But we don't raise the cwnd even though
the ACK does not indicate any congestion.
To fix this problem we should first call tcp_fastretrans_alert() to
process the dubious ACK and update the congestion state, then call
tcp_may_raise_cwnd() that raises cwnd based on the current state.
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: Nandita Dukkipati <nanditad@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>
PRR slow start is often too aggressive especially when drops are
caused by traffic policers. The policers mainly use token bucket
to enforce the rate so sending (twice) faster than the delivery
rate causes excessive drops.
This patch changes PRR to the conservative reduction bound
(CRB) mode in RFC 6937 by default. CRB follows the packet
conservation rule to send at most the delivery rate by default.
But if many packets are lost and the pipe is empty, CRB may take N
round trips to repair N losses. We conditionally turn on slow start
mode if all these conditions are made to speed up the recovery:
1) on the second round or later in recovery
2) retransmission sent in the previous round is delivered on this ACK
3) no retransmission is marked lost on this ACK
By using packet conservation by default, this change reduces the loss
retransmits signicantly on networks that deploy traffic policers,
up to 20% reduction of overall loss rate.
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: David S. Miller <davem@davemloft.net>
If the retransmission in CA_Loss is lost again, we should not
continue to slow start or raise cwnd in congestion avoidance mode.
Instead we should enter fast recovery and use PRR to reduce cwnd,
following the principle in RFC5681:
"... or the loss of a retransmission, should be taken as two
indications of congestion and, therefore, cwnd (and ssthresh) MUST
be lowered twice in this case."
This is especially important to reduce loss when the CA_Loss
state was caused by a traffic policer dropping the entire inflight.
The CA_Loss state has a problem where a loss of L packets causes the
sender to send a burst of L packets. So a policer that's dropping
most packets in a given RTT can cause a huge retransmit storm. By
contrast, PRR includes logic to bound the number of outbound packets
that result from a given ACK. So switching to CA_Recovery on lost
retransmits in CA_Loss avoids this retransmit storm problem when
in CA_Loss.
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: 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>
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>
Upcoming tcp_cdg uses tcp_enter_cwr() to initiate PRR. Export this
function so that CDG can be compiled as a module.
Cc: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Stephen Hemminger <stephen@networkplumber.org>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: David Hayes <davihay@ifi.uio.no>
Cc: Andreas Petlund <apetlund@simula.no>
Cc: Dave Taht <dave.taht@bufferbloat.net>
Cc: Nicolas Kuhn <nicolas.kuhn@telecom-bretagne.eu>
Signed-off-by: Kenneth Klette Jonassen <kennetkl@ifi.uio.no>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Conflicts:
drivers/net/ethernet/cadence/macb.c
drivers/net/phy/phy.c
include/linux/skbuff.h
net/ipv4/tcp.c
net/switchdev/switchdev.c
Switchdev was a case of RTNH_H_{EXTERNAL --> OFFLOAD}
renaming overlapping with net-next changes of various
sorts.
phy.c was a case of two changes, one adding a local
variable to a function whilst the second was removing
one.
tcp.c overlapped a deadlock fix with the addition of new tcp_info
statistic values.
macb.c involved the addition of two zyncq device entries.
skbuff.h involved adding back ipv4_daddr to nf_bridge_info
whilst net-next changes put two other existing members of
that struct into a union.
Signed-off-by: David S. Miller <davem@davemloft.net>
Taking socket spinlock in tcp_get_info() can deadlock, as
inet_diag_dump_icsk() holds the &hashinfo->ehash_locks[i],
while packet processing can use the reverse locking order.
We could avoid this locking for TCP_LISTEN states, but lockdep would
certainly get confused as all TCP sockets share same lockdep classes.
[ 523.722504] ======================================================
[ 523.728706] [ INFO: possible circular locking dependency detected ]
[ 523.734990] 4.1.0-dbg-DEV #1676 Not tainted
[ 523.739202] -------------------------------------------------------
[ 523.745474] ss/18032 is trying to acquire lock:
[ 523.750002] (slock-AF_INET){+.-...}, at: [<ffffffff81669d44>] tcp_get_info+0x2c4/0x360
[ 523.758129]
[ 523.758129] but task is already holding lock:
[ 523.763968] (&(&hashinfo->ehash_locks[i])->rlock){+.-...}, at: [<ffffffff816bcb75>] inet_diag_dump_icsk+0x1d5/0x6c0
[ 523.774661]
[ 523.774661] which lock already depends on the new lock.
[ 523.774661]
[ 523.782850]
[ 523.782850] the existing dependency chain (in reverse order) is:
[ 523.790326]
-> #1 (&(&hashinfo->ehash_locks[i])->rlock){+.-...}:
[ 523.796599] [<ffffffff811126bb>] lock_acquire+0xbb/0x270
[ 523.802565] [<ffffffff816f5868>] _raw_spin_lock+0x38/0x50
[ 523.808628] [<ffffffff81665af8>] __inet_hash_nolisten+0x78/0x110
[ 523.815273] [<ffffffff816819db>] tcp_v4_syn_recv_sock+0x24b/0x350
[ 523.822067] [<ffffffff81684d41>] tcp_check_req+0x3c1/0x500
[ 523.828199] [<ffffffff81682d09>] tcp_v4_do_rcv+0x239/0x3d0
[ 523.834331] [<ffffffff816842fe>] tcp_v4_rcv+0xa8e/0xc10
[ 523.840202] [<ffffffff81658fa3>] ip_local_deliver_finish+0x133/0x3e0
[ 523.847214] [<ffffffff81659a9a>] ip_local_deliver+0xaa/0xc0
[ 523.853440] [<ffffffff816593b8>] ip_rcv_finish+0x168/0x5c0
[ 523.859624] [<ffffffff81659db7>] ip_rcv+0x307/0x420
Lets use u64_sync infrastructure instead. As a bonus, 64bit
arches get optimized, as these are nop for them.
Fixes: 0df48c26d8 ("tcp: add tcpi_bytes_acked to tcp_info")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
After sending the new data packets to probe (step 2), F-RTO may
incorrectly send more probes if the next ACK advances SND_UNA and
does not sack new packet. However F-RTO RFC 5682 probes at most
once. This bug may cause sender to always send new data instead of
repairing holes, inducing longer HoL blocking on the receiver for
the application.
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>
Undo based on TCP timestamps should only happen on ACKs that advance
SND_UNA, according to the Eifel algorithm in RFC 3522:
Section 3.2:
(4) If the value of the Timestamp Echo Reply field of the
acceptable ACK's Timestamps option is smaller than the
value of RetransmitTS, then proceed to step (5),
Section Terminology:
We use the term 'acceptable ACK' as defined in [RFC793]. That is an
ACK that acknowledges previously unacknowledged data.
This is because upon receiving an out-of-order packet, the receiver
returns the last timestamp that advances RCV_NXT, not the current
timestamp of the packet in the DUPACK. Without checking the flag,
the DUPACK will cause tcp_packet_delayed() to return true and
tcp_try_undo_loss() will revert cwnd reduction.
Note that we check the condition in CA_Recovery already by only
calling tcp_try_undo_partial() if FLAG_SND_UNA_ADVANCED is set or
tcp_try_undo_recovery() if snd_una crosses high_seq.
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>
While testing tight tcp_mem settings, I found tcp sessions could be
stuck because we do not allow even one skb to be received on them.
By allowing one skb to be received, we introduce fairness and
eventuallu force memory hogs to release their allocation.
Signed-off-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>
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>
This patch allows a server application to get the TCP SYN headers for
its passive connections. This is useful if the server is doing
fingerprinting of clients based on SYN packet contents.
Two socket options are added: TCP_SAVE_SYN and TCP_SAVED_SYN.
The first is used on a socket to enable saving the SYN headers
for child connections. This can be set before or after the listen()
call.
The latter is used to retrieve the SYN headers for passive connections,
if the parent listener has enabled TCP_SAVE_SYN.
TCP_SAVED_SYN is read once, it frees the saved SYN headers.
The data returned in TCP_SAVED_SYN are network (IPv4/IPv6) and TCP
headers.
Original patch was written by Tom Herbert, I changed it to not hold
a full skb (and associated dst and conntracking reference).
We have used such patch for about 3 years at Google.
Signed-off-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>
Invoking pkts_acked is currently conditioned on FLAG_ACKED:
receiving a cumulative ACK of new data, or ACK with SYN flag set.
Remove this condition so that CC may get RTT measurements from all SACKs.
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Kenneth Klette Jonassen <kennetkl@ifi.uio.no>
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_sacktag_one() always picks the earliest sequence SACKed for RTT.
This might not make sense for congestion control in cases where:
1. ACKs are lost, i.e. a SACK following a lost SACK covers both
new and old segments at the receiver.
2. The receiver disregards the RFC 5681 recommendation to immediately
ACK out-of-order segments.
Give congestion control a RTT for the latest segment SACKed, which is the
most accurate RTT estimate, but preserve the conservative RTT for RTO.
Removes the call to skb_mstamp_get() in tcp_sacktag_one().
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Eric Dumazet <edumazet@google.com>
Signed-off-by: Kenneth Klette Jonassen <kennetkl@ifi.uio.no>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Later patch passes two values set in tcp_sacktag_one() to
tcp_clean_rtx_queue(). Prepare passing them via struct tcp_sacktag_state.
Acked-by: Yuchung Cheng <ycheng@google.com>
Cc: Eric Dumazet <edumazet@google.com>
Signed-off-by: Kenneth Klette Jonassen <kennetkl@ifi.uio.no>
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_mark_lost_retrans is not used when FACK is disabled. Since
tcp_update_reordering may disable FACK, it should be called first
before tcp_mark_lost_retrans.
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: David S. Miller <davem@davemloft.net>
This patch tracks total number of payload bytes received on a TCP socket.
This is the sum of all changes done to tp->rcv_nxt
RFC4898 named this : tcpEStatsAppHCThruOctetsReceived
This is a 64bit field, and can be fetched both from TCP_INFO
getsockopt() if one has a handle on a TCP socket, or from inet_diag
netlink facility (iproute2/ss patch will follow)
Note that tp->bytes_received was placed near tp->rcv_nxt for
best data locality and minimal performance impact.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Matt Mathis <mattmathis@google.com>
Cc: Eric Salo <salo@google.com>
Cc: Martin Lau <kafai@fb.com>
Cc: Chris Rapier <rapier@psc.edu>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch tracks total number of bytes acked for a TCP socket.
This is the sum of all changes done to tp->snd_una, and allows
for precise tracking of delivered data.
RFC4898 named this : tcpEStatsAppHCThruOctetsAcked
This is a 64bit field, and can be fetched both from TCP_INFO
getsockopt() if one has a handle on a TCP socket, or from inet_diag
netlink facility (iproute2/ss patch will follow)
Note that tp->bytes_acked was placed near tp->snd_una for
best data locality and minimal performance impact.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Cc: Matt Mathis <mattmathis@google.com>
Cc: Eric Salo <salo@google.com>
Cc: Martin Lau <kafai@fb.com>
Cc: Chris Rapier <rapier@psc.edu>
Signed-off-by: David S. Miller <davem@davemloft.net>
Ensure that we either see that the buffer has write space
in tcp_poll() or that we perform a wakeup from the input
side. Did not run into any actual problem here, but thought
that we should make things explicit.
Signed-off-by: Jason Baron <jbaron@akamai.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Since retransmitted segments are not used for RTT estimation, previously
SACKed segments present in the rtx queue are used. This estimation can be
several times larger than the actual RTT. When a cumulative ack covers both
previously SACKed and retransmitted segments, CC may thus get a bogus RTT.
Such segments previously had an RTT estimation in tcp_sacktag_one(), so it
seems reasonable to not reuse them in tcp_clean_rtx_queue() at all.
Afaik, this has had no effect on SRTT/RTO because of Karn's check.
Signed-off-by: Kenneth Klette Jonassen <kennetkl@ifi.uio.no>
Acked-by: Neal Cardwell <ncardwell@google.com>
Tested-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Yuchung Cheng <ycheng@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>
Conflicts:
drivers/net/ethernet/mellanox/mlx4/cmd.c
net/core/fib_rules.c
net/ipv4/fib_frontend.c
The fib_rules.c and fib_frontend.c conflicts were locking adjustments
in 'net' overlapping addition and removal of code in 'net-next'.
The mlx4 conflict was a bug fix in 'net' happening in the same
place a constant was being replaced with a more suitable macro.
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>
On processing cumulative ACKs, the FRTO code was not checking the
SACKed bit, meaning that there could be a spurious FRTO undo on a
cumulative ACK of a previously SACKed skb.
The FRTO code should only consider a cumulative ACK to indicate that
an original/unretransmitted skb is newly ACKed if the skb was not yet
SACKed.
The effect of the spurious FRTO undo would typically be to make the
connection think that all previously-sent packets were in flight when
they really weren't, leading to a stall and an RTO.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Fixes: e33099f96d ("tcp: implement RFC5682 F-RTO")
Signed-off-by: David S. Miller <davem@davemloft.net>
After commit 1fb6f159fd ("tcp: add tcp_conn_request"),
tcp_syn_flood_action() is no longer used from IPv6.
We can make it static, by moving it above tcp_conn_request()
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Octavian Purdila <octavian.purdila@intel.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
ss should display ipv4 mapped request sockets like this :
tcp SYN-RECV 0 0 ::ffff:192.168.0.1:8080 ::ffff:192.0.2.1:35261
and not like this :
tcp SYN-RECV 0 0 192.168.0.1:8080 192.0.2.1:35261
We should init ireq->ireq_family based on listener sk_family,
not the actual protocol carried by SYN packet.
This means we can set ireq_family in inet_reqsk_alloc()
Fixes: 3f66b083a5 ("inet: introduce ireq_family")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
When request sock are put in ehash table, the whole notion
of having a previous request to update dl_next is pointless.
Also, following patch will get rid of big purge timer,
so we want to delete a request sock without holding listener lock.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
While testing last patch series, I found req sock refcounting was wrong.
We must set skc_refcnt to 1 for all request socks added in hashes,
but also on request sockets created by FastOpen or syncookies.
It is tricky because we need to defer this initialization so that
future RCU lookups do not try to take a refcount on a not yet
fully initialized request socket.
Also get rid of ireq_refcnt alias.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Fixes: 13854e5a60 ("inet: add proper refcounting to request sock")
Signed-off-by: David S. Miller <davem@davemloft.net>
The listener field in struct tcp_request_sock is a pointer
back to the listener. We now have req->rsk_listener, so TCP
only needs one boolean and not a full pointer.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Once we'll be able to lookup request sockets in ehash table,
we'll need to get access to listener which created this request.
This avoid doing a lookup to find the listener, which benefits
for a more solid SO_REUSEPORT, and is needed once we no
longer queue request sock into a listener private queue.
Note that 'struct tcp_request_sock'->listener could be reduced
to a single bit, as TFO listener should match req->rsk_listener.
TFO will no longer need to hold a reference on the listener.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
inet_reqsk_alloc() is becoming fat and should not be inlined.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
listener socket can be used to set net pointer, and will
be later used to hold a reference on listener.
Add a const qualifier to first argument (struct request_sock_ops *),
and factorize all write_pnet(&ireq->ireq_net, sock_net(sk));
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_oow_rate_limited() is hardly used in fast path, there is
no point inlining it.
Signed-of-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This big helper is called once from tcp_conn_request(), there is no
point having it in an include. Compiler will inline it anyway.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Once request socks will be in ehash table, they will need to have
a valid ir_iff field.
This is currently true only for IPv6. This patch extends support
for IPv4 as well.
This means inet_diag_fill_req() can now properly use ir_iif,
which is better for IPv6 link locals anyway, as request sockets
and established sockets will propagate consistent netlink idiag_if.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
I forgot to update dccp_v6_conn_request() & cookie_v6_check().
They both need to set ireq->ireq_net and ireq->ir_cookie
Lets clear ireq->ir_cookie in inet_reqsk_alloc()
Signed-off-by: Eric Dumazet <edumazet@google.com>
Fixes: 33cf7c90fe ("net: add real socket cookies")
Signed-off-by: David S. Miller <davem@davemloft.net>
I forgot to use write_pnet() in three locations.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Fixes: 33cf7c90fe ("net: add real socket cookies")
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
A long standing problem in netlink socket dumps is the use
of kernel socket addresses as cookies.
1) It is a security concern.
2) Sockets can be reused quite quickly, so there is
no guarantee a cookie is used once and identify
a flow.
3) request sock, establish sock, and timewait socks
for a given flow have different cookies.
Part of our effort to bring better TCP statistics requires
to switch to a different allocator.
In this patch, I chose to use a per network namespace 64bit generator,
and to use it only in the case a socket needs to be dumped to netlink.
(This might be refined later if needed)
Note that I tried to carry cookies from request sock, to establish sock,
then timewait sockets.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Eric Salo <salo@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_should_expand_sndbuf() does not expand the send buffer if we have
filled the congestion window.
However, it should use tcp_packets_in_flight() instead of
tp->packets_out to make this check.
Testing has established that the difference matters a lot if there are
many SACKed packets, causing a needless performance shortfall.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
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: David S. Miller <davem@davemloft.net>
Ensure that in state ESTABLISHED, where the connection is represented
by a tcp_sock, we rate limit dupacks in response to incoming packets
(a) with TCP timestamps that fail PAWS checks, or (b) with sequence
numbers or ACK numbers that are out of the acceptable window.
We do not send a dupack in response to out-of-window packets if it has
been less than sysctl_tcp_invalid_ratelimit (default 500ms) since we
last sent a dupack in response to an out-of-window packet.
There is already a similar (although global) rate-limiting mechanism
for "challenge ACKs". When deciding whether to send a challence ACK,
we first consult the new per-connection rate limit, and then the
global rate limit.
Reported-by: Avery Fay <avery@mixpanel.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Helpers for mitigating ACK loops by rate-limiting dupacks sent in
response to incoming out-of-window packets.
This patch includes:
- rate-limiting logic
- sysctl to control how often we allow dupacks to out-of-window packets
- SNMP counter for cases where we rate-limited our dupack sending
The rate-limiting logic in this patch decides to not send dupacks in
response to out-of-window segments if (a) they are SYNs or pure ACKs
and (b) the remote endpoint is sending them faster than the configured
rate limit.
We rate-limit our responses rather than blocking them entirely or
resetting the connection, because legitimate connections can rely on
dupacks in response to some out-of-window segments. For example, zero
window probes are typically sent with a sequence number that is below
the current window, and ZWPs thus expect to thus elicit a dupack in
response.
We allow dupacks in response to TCP segments with data, because these
may be spurious retransmissions for which the remote endpoint wants to
receive DSACKs. This is safe because segments with data can't
realistically be part of ACK loops, which by their nature consist of
each side sending pure/data-less ACKs to each other.
The dupack interval is controlled by a new sysctl knob,
tcp_invalid_ratelimit, given in milliseconds, in case an administrator
needs to dial this upward in the face of a high-rate DoS attack. The
name and units are chosen to be analogous to the existing analogous
knob for ICMP, icmp_ratelimit.
The default value for tcp_invalid_ratelimit is 500ms, which allows at
most one such dupack per 500ms. This is chosen to be 2x faster than
the 1-second minimum RTO interval allowed by RFC 6298 (section 2, rule
2.4). We allow the extra 2x factor because network delay variations
can cause packets sent at 1 second intervals to be compressed and
arrive much closer.
Reported-by: Avery Fay <avery@mixpanel.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
One deployment requirement of DCTCP is to be able to run
in a DC setting along with TCP traffic. As Glenn Judd's
NSDI'15 paper "Attaining the Promise and Avoiding the Pitfalls
of TCP in the Datacenter" [1] (tba) explains, one way to
solve this on switch side is to split DCTCP and TCP traffic
in two queues per switch port based on the DSCP: one queue
soley intended for DCTCP traffic and one for non-DCTCP traffic.
For the DCTCP queue, there's the marking threshold K as
explained in commit e3118e8359 ("net: tcp: add DCTCP congestion
control algorithm") for RED marking ECT(0) packets with CE.
For the non-DCTCP queue, there's f.e. a classic tail drop queue.
As already explained in e3118e8359, running DCTCP at scale
when not marking SYN/SYN-ACK packets with ECT(0) has severe
consequences as for non-ECT(0) packets, traversing the RED
marking DCTCP queue will result in a severe reduction of
connection probability.
This is due to the DCTCP queue being dominated by ECT(0) traffic
and switches handle non-ECT traffic in the RED marking queue
after passing K as drops, where K is usually a low watermark
in order to leave enough tailroom for bursts. Splitting DCTCP
traffic among several queues (ECN and non-ECN queue) is being
considered a terrible idea in the network community as it
splits single flows across multiple network paths.
Therefore, commit e3118e8359 implements this on Linux as
ECT(0) marked traffic, as we argue that marking all packets
of a DCTCP flow is the only viable solution and also doesn't
speak against the draft.
However, recently, a DCTCP implementation for FreeBSD hit also
their mainline kernel [2]. In order to let them play well
together with Linux' DCTCP, we would need to loosen the
requirement that ECT(0) has to be asserted during the 3WHS as
not implemented in FreeBSD. This simplifies the ECN test and
lets DCTCP work together with FreeBSD.
Joint work with Daniel Borkmann.
[1] https://www.usenix.org/conference/nsdi15/technical-sessions/presentation/judd
[2] 8ad8794452
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
Cc: Glenn Judd <glenn.judd@morganstanley.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Current behavior only passes RTTs from sequentially acked data to CC.
If sender gets a combined ACK for segment 1 and SACK for segment 3, then the
computed RTT for CC is the time between sending segment 1 and receiving SACK
for segment 3.
Pass the minimum computed RTT from any acked data to CC, i.e. time between
sending segment 3 and receiving SACK for segment 3.
Signed-off-by: Kenneth Klette Jonassen <kennetkl@ifi.uio.no>
Signed-off-by: David S. Miller <davem@davemloft.net>
With TLP, the peer may reply to a probe with an
ACK+D-SACK, with ack value set to tlp_high_seq. In the current code,
such ACK+DSACK will be missed and only at next, higher ack will the TLP
episode be considered done. Since the DSACK is not present anymore,
this will cost a cwnd reduction.
This patch ensures that this scenario does not cause a cwnd reduction, since
receiving an ACK+DSACK indicates that both the initial segment and the probe
have been received by the peer.
The following packetdrill test, from Neal Cardwell, validates this patch:
// Establish a connection.
0 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3
+0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0
+0 bind(3, ..., ...) = 0
+0 listen(3, 1) = 0
+0 < S 0:0(0) win 32792 <mss 1000,sackOK,nop,nop,nop,wscale 7>
+0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6>
+.020 < . 1:1(0) ack 1 win 257
+0 accept(3, ..., ...) = 4
// Send 1 packet.
+0 write(4, ..., 1000) = 1000
+0 > P. 1:1001(1000) ack 1
// Loss probe retransmission.
// packets_out == 1 => schedule PTO in max(2*RTT, 1.5*RTT + 200ms)
// In this case, this means: 1.5*RTT + 200ms = 230ms
+.230 > P. 1:1001(1000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10 }%
// Receiver ACKs at tlp_high_seq with a DSACK,
// indicating they received the original packet and probe.
+.020 < . 1:1(0) ack 1001 win 257 <sack 1:1001,nop,nop>
+0 %{ assert tcpi_snd_cwnd == 10 }%
// Send another packet.
+0 write(4, ..., 1000) = 1000
+0 > P. 1001:2001(1000) ack 1
// Receiver ACKs above tlp_high_seq, which should end the TLP episode
// if we haven't already. We should not reduce cwnd.
+.020 < . 1:1(0) ack 2001 win 257
+0 %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }%
Credits:
-Gregory helped in finding that tcp_process_tlp_ack was where the cwnd
got reduced in our MPTCP tests.
-Neal wrote the packetdrill test above
-Yuchung reworked the patch to make it more readable.
Cc: Gregory Detal <gregory.detal@uclouvain.be>
Cc: Nandita Dukkipati <nanditad@google.com>
Tested-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Yuchung Cheng <ycheng@google.com>
Reviewed-by: Eric Dumazet <eric.dumazet@gmail.com>
Signed-off-by: Sébastien Barré <sebastien.barre@uclouvain.be>
Acked-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
