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: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil 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: NDavid 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: c0402760 ("tcp: new CC hook to set sending rate with rate_sample in any CA state")
Signed-off-by: NWei Wang <weiwan@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Whole point of randomization was to hide server uptime, but an attacker
can simply start a syn flood and TCP generates 'old style' timestamps,
directly revealing server jiffies value.

Also, TSval sent by the server to a particular remote address vary
depending on syncookies being sent or not, potentially triggering PAWS
drops for innocent clients.

Lets implement proper randomization, including for SYNcookies.

Also we do not need to export sysctl_tcp_timestamps, since it is not
used from a module.

In v2, I added Florian feedback and contribution, adding tsoff to
tcp_get_cookie_sock().

v3 removed one unused variable in tcp_v4_connect() as Florian spotted.

Fixes: 95a22cae ("tcp: randomize tcp timestamp offsets for each connection")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reviewed-by: NFlorian Westphal <fw@strlen.de>
Tested-by: NFlorian Westphal <fw@strlen.de>
Cc: Yuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

No longer needed, since tp->tcp_mstamp holds the information.

This is needed to remove sack_state.ack_time in a following patch.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

No longer needed, since tp->tcp_mstamp holds the information.

This is needed to remove sack_state.ack_time in a following patch.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This is no longer used, since tcp_rack_detect_loss() takes
the timestamp from tp->tcp_mstamp
Signed-off-by: NEric Dumazet <edumazet@google.com>
Acked-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This counter records the number of times the firewall blackhole issue is
detected and active TFO is disabled.
Signed-off-by: NWei Wang <weiwan@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Middlebox firewall issues can potentially cause server's data being
blackholed after a successful 3WHS using TFO. Following are the related
reports from Apple:
https://www.nanog.org/sites/default/files/Paasch_Network_Support.pdf
Slide 31 identifies an issue where the client ACK to the server's data
sent during a TFO'd handshake is dropped.
C ---> syn-data ---> S
C <--- syn/ack ----- S
C (accept & write)
C <---- data ------- S
C ----- ACK -> X     S
		[retry and timeout]

https://www.ietf.org/proceedings/94/slides/slides-94-tcpm-13.pdf
Slide 5 shows a similar situation that the server's data gets dropped
after 3WHS.
C ---- syn-data ---> S
C <--- syn/ack ----- S
C ---- ack --------> S
S (accept & write)
C?  X <- data ------ S
		[retry and timeout]

This is the worst failure b/c the client can not detect such behavior to
mitigate the situation (such as disabling TFO). Failing to proceed, the
application (e.g., SSL library) may simply timeout and retry with TFO
again, and the process repeats indefinitely.

The proposed solution is to disable active TFO globally under the
following circumstances:
1. client side TFO socket detects out of order FIN
2. client side TFO socket receives out of order RST

We disable active side TFO globally for 1hr at first. Then if it
happens again, we disable it for 2h, then 4h, 8h, ...
And we reset the timeout to 1hr if a client side TFO sockets not opened
on loopback has successfully received data segs from server.
And we examine this condition during close().

The rational behind it is that when such firewall issue happens,
application running on the client should eventually close the socket as
it is not able to get the data it is expecting. Or application running
on the server should close the socket as it is not able to receive any
response from client.
In both cases, out of order FIN or RST will get received on the client
given that the firewall will not block them as no data are in those
frames.
And we want to disable active TFO globally as it helps if the middle box
is very close to the client and most of the connections are likely to
fail.

Also, add a debug sysctl:
  tcp_fastopen_blackhole_detect_timeout_sec:
    the initial timeout to use when firewall blackhole issue happens.
    This can be set and read.
    When setting it to 0, it means to disable the active disable logic.
Signed-off-by: NWei Wang <weiwan@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid 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: NGao Feng <fgao@ikuai8.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Because sysctl_tcp_adv_win_scale could be changed any time, so there
is one race in tcp_win_from_space.
For example,
1.sysctl_tcp_adv_win_scale<=0 (sysctl_tcp_adv_win_scale is negative now)
2.space>>(-sysctl_tcp_adv_win_scale) (sysctl_tcp_adv_win_scale is postive now)

As a result, tcp_win_from_space returns 0. It is unexpected.

Certainly if the compiler put the sysctl_tcp_adv_win_scale into one
register firstly, then use the register directly, it would be ok.
But we could not depend on the compiler behavior.
Signed-off-by: NGao Feng <fgao@ikuai8.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The tcp_tw_recycle was already broken for connections
behind NAT, since the per-destination timestamp is not
monotonically increasing for multiple machines behind
a single destination address.

After the randomization of TCP timestamp offsets
in commit 8a5bd45f6616 (tcp: randomize tcp timestamp offsets
for each connection), the tcp_tw_recycle is broken for all
types of connections for the same reason: the timestamps
received from a single machine is not monotonically increasing,
anymore.

Remove tcp_tw_recycle, since it is not functional. Also, remove
the PAWSPassive SNMP counter since it is only used for
tcp_tw_recycle, and simplify tcp_v4_route_req and tcp_v6_route_req
since the strict argument is only set when tcp_tw_recycle is
enabled.
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Cc: Lutz Vieweg <lvml@5t9.de>
Cc: Florian Westphal <fw@strlen.de>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Commit 8a5bd45f6616 (tcp: randomize tcp timestamp offsets for each connection)
randomizes TCP timestamps per connection. After this commit,
there is no guarantee that the timestamps received from the
same destination are monotonically increasing. As a result,
the per-destination timestamp cache in TCP metrics (i.e., tcpm_ts
in struct tcp_metrics_block) is broken and cannot be relied upon.

Remove the per-destination timestamp cache and all related code
paths.

Note that this cache was already broken for caching timestamps of
multiple machines behind a NAT sharing the same address.
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Cc: Lutz Vieweg <lvml@5t9.de>
Cc: Florian Westphal <fw@strlen.de>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The functions that are returning tcp sequence number also setup
TS offset value, so rename them to better describe their purpose.

No functional changes in this patch.
Suggested-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NAlexey Kodanev <alexey.kodanev@oracle.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch adds a new socket option, TCP_FASTOPEN_CONNECT, as an
alternative way to perform Fast Open on the active side (client). Prior
to this patch, a client needs to replace the connect() call with
sendto(MSG_FASTOPEN). This can be cumbersome for applications who want
to use Fast Open: these socket operations are often done in lower layer
libraries used by many other applications. Changing these libraries
and/or the socket call sequences are not trivial. A more convenient
approach is to perform Fast Open by simply enabling a socket option when
the socket is created w/o changing other socket calls sequence:
  s = socket()
    create a new socket
  setsockopt(s, IPPROTO_TCP, TCP_FASTOPEN_CONNECT …);
    newly introduced sockopt
    If set, new functionality described below will be used.
    Return ENOTSUPP if TFO is not supported or not enabled in the
    kernel.

  connect()
    With cookie present, return 0 immediately.
    With no cookie, initiate 3WHS with TFO cookie-request option and
    return -1 with errno = EINPROGRESS.

  write()/sendmsg()
    With cookie present, send out SYN with data and return the number of
    bytes buffered.
    With no cookie, and 3WHS not yet completed, return -1 with errno =
    EINPROGRESS.
    No MSG_FASTOPEN flag is needed.

  read()
    Return -1 with errno = EWOULDBLOCK/EAGAIN if connect() is called but
    write() is not called yet.
    Return -1 with errno = EWOULDBLOCK/EAGAIN if connection is
    established but no msg is received yet.
    Return number of bytes read if socket is established and there is
    msg received.

The new API simplifies life for applications that always perform a write()
immediately after a successful connect(). Such applications can now take
advantage of Fast Open by merely making one new setsockopt() call at the time
of creating the socket. Nothing else about the application's socket call
sequence needs to change.
Signed-off-by: NWei Wang <weiwan@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid 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: NWei Wang <weiwan@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid 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: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This patch changes two things:

1. Start fast recovery with RACK in addition to other heuristics
   (e.g., DUPACK threshold, FACK). Prior to this change RACK
   is enabled to detect losses only after the recovery has
   started by other algorithms.

2. Disable TCP early retransmit. RACK subsumes the early retransmit
   with the new reordering timer feature. A latter patch in this
   series removes the early retransmit code.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The packets inside a jumbo skb (e.g., TSO) share the same skb
timestamp, even though they are sent sequentially on the wire. Since
RACK is based on time, it can not detect some packets inside the
same skb are lost.  However, we can leverage the packet sequence
numbers as extended timestamps to detect losses. Therefore, when
RACK timestamp is identical to skb's timestamp (i.e., one of the
packets of the skb is acked or sacked), we use the sequence numbers
of the acked and unacked packets to break ties.

We can use the same sequence logic to advance RACK xmit time as
well to detect more losses and avoid timeout.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid 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: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Record the most recent RTT in RACK. It is often identical to the
"ca_rtt_us" values in tcp_clean_rtx_queue. But when the packet has
been retransmitted, RACK choses to believe the ACK is for the
(latest) retransmitted packet if the RTT is over minimum RTT.

This requires passing the arrival time of the most recent ACK to
RACK routines. The timestamp is now recorded in the "ack_time"
in tcp_sacktag_state during the ACK processing.

This patch does not change the RACK algorithm itself. It only adds
the RTT variable to prepare the next main patch.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Create a new helper tcp_rack_detect_loss to prepare the upcoming
RACK reordering timer patch.
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Acked-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Different namespace application might require fast recycling
TIME-WAIT sockets independently of the host.
Signed-off-by: NHaishuang Yan <yanhaishuang@cmss.chinamobile.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Different namespaces might have different requirements to reuse
TIME-WAIT sockets for new connections. This might be required in
cases where different namespace applications are in place which
require TIME_WAIT socket connections to be reduced independently
of the host.
Signed-off-by: NHaishuang Yan <yanhaishuang@cmss.chinamobile.com>
Signed-off-by: NDavid 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 ceaa1fef ("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: NFlorian Westphal <fw@strlen.de>
Acked-by: NEric Dumazet <edumazet@google.com>
Acked-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid 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: NFrancis Yan <francisyyan@gmail.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid 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: NFrancis Yan <francisyyan@gmail.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The undo_cwnd fallback in the stack doubles cwnd based on ssthresh,
which un-does reno halving behaviour.

It seems more appropriate to let congctl algorithms pair .ssthresh
and .undo_cwnd properly. Add a 'tcp_reno_undo_cwnd' function and wire it
up for all congestion algorithms that used to rely on the fallback.

Cc: Eric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Signed-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

With syzkaller help, Marco Grassi found a bug in TCP stack,
crashing in tcp_collapse()

Root cause is that sk_filter() can truncate the incoming skb,
but TCP stack was not really expecting this to happen.
It probably was expecting a simple DROP or ACCEPT behavior.

We first need to make sure no part of TCP header could be removed.
Then we need to adjust TCP_SKB_CB(skb)->end_seq

Many thanks to syzkaller team and Marco for giving us a reproducer.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Reported-by: NMarco Grassi <marco.gra@gmail.com>
Reported-by: NVladis Dronov <vdronov@redhat.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Andrey reported the following error report while running the syzkaller
fuzzer:

general protection fault: 0000 [#1] SMP KASAN
Dumping ftrace buffer:
   (ftrace buffer empty)
Modules linked in:
CPU: 0 PID: 648 Comm: syz-executor Not tainted 4.9.0-rc3+ #333
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011
task: ffff8800398c4480 task.stack: ffff88003b468000
RIP: 0010:[<ffffffff83091106>]  [<     inline     >]
inet_exact_dif_match include/net/tcp.h:808
RIP: 0010:[<ffffffff83091106>]  [<ffffffff83091106>]
__inet_lookup_listener+0xb6/0x500 net/ipv4/inet_hashtables.c:219
RSP: 0018:ffff88003b46f270  EFLAGS: 00010202
RAX: 0000000000000004 RBX: 0000000000004242 RCX: 0000000000000001
RDX: 0000000000000000 RSI: ffffc90000e3c000 RDI: 0000000000000054
RBP: ffff88003b46f2d8 R08: 0000000000004000 R09: ffffffff830910e7
R10: 0000000000000000 R11: 000000000000000a R12: ffffffff867fa0c0
R13: 0000000000004242 R14: 0000000000000003 R15: dffffc0000000000
FS:  00007fb135881700(0000) GS:ffff88003ec00000(0000) knlGS:0000000000000000
CS:  0010 DS: 0000 ES: 0000 CR0: 0000000080050033
CR2: 0000000020cc3000 CR3: 000000006d56a000 CR4: 00000000000006f0
Stack:
 0000000000000000 000000000601a8c0 0000000000000000 ffffffff00004242
 424200003b9083c2 ffff88003def4041 ffffffff84e7e040 0000000000000246
 ffff88003a0911c0 0000000000000000 ffff88003a091298 ffff88003b9083ae
Call Trace:
 [<ffffffff831100f4>] tcp_v4_send_reset+0x584/0x1700 net/ipv4/tcp_ipv4.c:643
 [<ffffffff83115b1b>] tcp_v4_rcv+0x198b/0x2e50 net/ipv4/tcp_ipv4.c:1718
 [<ffffffff83069d22>] ip_local_deliver_finish+0x332/0xad0
net/ipv4/ip_input.c:216
...

MD5 has a code path that calls __inet_lookup_listener with a null skb,
so inet{6}_exact_dif_match needs to check skb against null before pulling
the flag.

Fixes: a04a480d ("net: Require exact match for TCP socket lookups if
       dif is l3mdev")
Reported-by: NAndrey Konovalov <andreyknvl@google.com>
Signed-off-by: NDavid Ahern <dsa@cumulusnetworks.com>
Tested-by: NAndrey Konovalov <andreyknvl@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Currently, socket lookups for l3mdev (vrf) use cases can match a socket
that is bound to a port but not a device (ie., a global socket). If the
sysctl tcp_l3mdev_accept is not set this leads to ack packets going out
based on the main table even though the packet came in from an L3 domain.
The end result is that the connection does not establish creating
confusion for users since the service is running and a socket shows in
ss output. Fix by requiring an exact dif to sk_bound_dev_if match if the
skb came through an interface enslaved to an l3mdev device and the
tcp_l3mdev_accept is not set.

skb's through an l3mdev interface are marked by setting a flag in
inet{6}_skb_parm. The IPv6 variant is already set; this patch adds the
flag for IPv4. Using an skb flag avoids a device lookup on the dif. The
flag is set in the VRF driver using the IP{6}CB macros. For IPv4, the
inet_skb_parm struct is moved in the cb per commit 971f10ec, so the
match function in the TCP stack needs to use TCP_SKB_CB. For IPv6, the
move is done after the socket lookup, so IP6CB is used.

The flags field in inet_skb_parm struct needs to be increased to add
another flag. There is currently a 1-byte hole following the flags,
so it can be expanded to u16 without increasing the size of the struct.

Fixes: 193125db ("net: Introduce VRF device driver")
Signed-off-by: NDavid Ahern <dsa@cumulusnetworks.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This commit introduces an optional new "omnipotent" hook,
cong_control(), for congestion control modules. The cong_control()
function is called at the end of processing an ACK (i.e., after
updating sequence numbers, the SACK scoreboard, and loss
detection). At that moment we have precise delivery rate information
the congestion control module can use to control the sending behavior
(using cwnd, TSO skb size, and pacing rate) in any CA state.

This function can also be used by a congestion control that prefers
not to use the default cwnd reduction approach (i.e., the PRR
algorithm) during CA_Recovery to control the cwnd and sending rate
during loss recovery.

We take advantage of the fact that recent changes defer the
retransmission or transmission of new data (e.g. by F-RTO) in recovery
until the new tcp_cong_control() function is run.

With this commit, we only run tcp_update_pacing_rate() if the
congestion control is not using this new API. New congestion controls
which use the new API do not want the TCP stack to run the default
pacing rate calculation and overwrite whatever pacing rate they have
chosen at initialization time.
Signed-off-by: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Currently the TCP send buffer expands to twice cwnd, in order to allow
limited transmits in the CA_Recovery state. This assumes that cwnd
does not increase in the CA_Recovery.

For some congestion control algorithms, like the upcoming BBR module,
if the losses in recovery do not indicate congestion then we may
continue to raise cwnd multiplicatively in recovery. In such cases the
current multiplier will falsely limit the sending rate, much as if it
were limited by the application.

This commit adds an optional congestion control callback to use a
different multiplier to expand the TCP send buffer. For congestion
control modules that do not specificy this callback, TCP continues to
use the previous default of 2.
Signed-off-by: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NStephen Hemminger <stephen@networkplumber.org>
Signed-off-by: NDavid 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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid 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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

This commit adds code to track whether the delivery rate represented
by each rate_sample was limited by the application.

Upon each transmit, we store in the is_app_limited field in the skb a
boolean bit indicating whether there is a known "bubble in the pipe":
a point in the rate sample interval where the sender was
application-limited, and did not transmit even though the cwnd and
pacing rate allowed it.

This logic marks the flow app-limited on a write if *all* of the
following are true:

  1) There is less than 1 MSS of unsent data in the write queue
     available to transmit.

  2) There is no packet in the sender's queues (e.g. in fq or the NIC
     tx queue).

  3) The connection is not limited by cwnd.

  4) There are no lost packets to retransmit.

The tcp_rate_check_app_limited() code in tcp_rate.c determines whether
the connection is application-limited at the moment. If the flow is
application-limited, it sets the tp->app_limited field. If the flow is
application-limited then that means there is effectively a "bubble" of
silence in the pipe now, and this silence will be reflected in a lower
bandwidth sample for any rate samples from now until we get an ACK
indicating this bubble has exited the pipe: specifically, until we get
an ACK for the next packet we transmit.

When we send every skb we record in scb->tx.is_app_limited whether the
resulting rate sample will be application-limited.

The code in tcp_rate_gen() checks to see when it is safe to mark all
known application-limited bubbles of silence as having exited the
pipe. It does this by checking to see when the delivered count moves
past the tp->app_limited marker. At this point it zeroes the
tp->app_limited marker, as all known bubbles are out of the pipe.

We make room for the tx.is_app_limited bit in the skb by borrowing a
bit from the in_flight field used by NV to record the number of bytes
in flight. The receive window in the TCP header is 16 bits, and the
max receive window scaling shift factor is 14 (RFC 1323). So the max
receive window offered by the TCP protocol is 2^(16+14) = 2^30. So we
only need 30 bits for the tx.in_flight used by NV.
Signed-off-by: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid 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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Refactor the TCP min_rtt code to reuse the new win_minmax library in
lib/win_minmax.c to simplify the TCP code.

This is a pure refactor: the functionality is exactly the same. We
just moved the windowed min code to make TCP easier to read and
maintain, and to allow other parts of the kernel to use the windowed
min/max filter code.
Signed-off-by: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Over the years, TCP BDP has increased by several orders of magnitude,
and some people are considering to reach the 2 Gbytes limit.

Even with current window scale limit of 14, ~1 Gbytes maps to ~740,000
MSS.

In presence of packet losses (or reorders), TCP stores incoming packets
into an out of order queue, and number of skbs sitting there waiting for
the missing packets to be received can be in the 10^5 range.

Most packets are appended to the tail of this queue, and when
packets can finally be transferred to receive queue, we scan the queue
from its head.

However, in presence of heavy losses, we might have to find an arbitrary
point in this queue, involving a linear scan for every incoming packet,
throwing away cpu caches.

This patch converts it to a RB tree, to get bounded latencies.

Yaogong wrote a preliminary patch about 2 years ago.
Eric did the rebase, added ofo_last_skb cache, polishing and tests.

Tested with network dropping between 1 and 10 % packets, with good
success (about 30 % increase of throughput in stress tests)

Next step would be to also use an RB tree for the write queue at sender
side ;)
Signed-off-by: NYaogong Wang <wygivan@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Acked-By: NIlpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

When TCP operates in lossy environments (between 1 and 10 % packet
losses), many SACK blocks can be exchanged, and I noticed we could
drop them on busy senders, if these SACK blocks have to be queued
into the socket backlog.

While the main cause is the poor performance of RACK/SACK processing,
we can try to avoid these drops of valuable information that can lead to
spurious timeouts and retransmits.

Cause of the drops is the skb->truesize overestimation caused by :

- drivers allocating ~2048 (or more) bytes as a fragment to hold an
  Ethernet frame.

- various pskb_may_pull() calls bringing the headers into skb->head
  might have pulled all the frame content, but skb->truesize could
  not be lowered, as the stack has no idea of each fragment truesize.

The backlog drops are also more visible on bidirectional flows, since
their sk_rmem_alloc can be quite big.

Let's add some room for the backlog, as only the socket owner
can selectively take action to lower memory needs, like collapsing
receive queues or partial ofo pruning.
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Acked-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

In inet_stream_ops we set read_sock to tcp_read_sock and peek_len to
tcp_peek_len (which is just a stub function that calls tcp_inq).
Signed-off-by: NTom Herbert <tom@herbertland.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>