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  1. 12 12月, 2017 1 次提交
  3. 08 12月, 2017 1 次提交
  5. 11 11月, 2017 2 次提交
  7. 05 11月, 2017 1 次提交
    • P
      tcp: higher throughput under reordering with adaptive RACK reordering wnd · 1f255691
      由 Priyaranjan Jha 提交于 
      Currently TCP RACK loss detection does not work well if packets are
being reordered beyond its static reordering window (min_rtt/4).Under
such reordering it may falsely trigger loss recoveries and reduce TCP
throughput significantly.

This patch improves that by increasing and reducing the reordering
window based on DSACK, which is now supported in major TCP implementations.
It makes RACK's reo_wnd adaptive based on DSACK and no. of recoveries.

- If DSACK is received, increment reo_wnd by min_rtt/4 (upper bounded
  by srtt), since there is possibility that spurious retransmission was
  due to reordering delay longer than reo_wnd.

- Persist the current reo_wnd value for TCP_RACK_RECOVERY_THRESH (16)
  no. of successful recoveries (accounts for full DSACK-based loss
  recovery undo). After that, reset it to default (min_rtt/4).

- At max, reo_wnd is incremented only once per rtt. So that the new
  DSACK on which we are reacting, is due to the spurious retx (approx)
  after the reo_wnd has been updated last time.

- reo_wnd is tracked in terms of steps (of min_rtt/4), rather than
  absolute value to account for change in rtt.

In our internal testing, we observed significant increase in throughput,
in scenarios where reordering exceeds min_rtt/4 (previous static value).
Signed-off-by: NPriyaranjan Jha <priyarjha@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>
      1f255691
  9. 26 10月, 2017 1 次提交
  11. 24 10月, 2017 1 次提交
    • C
      tcp: Configure TFO without cookie per socket and/or per route · 71c02379
      由 Christoph Paasch 提交于 
      We already allow to enable TFO without a cookie by using the
fastopen-sysctl and setting it to TFO_SERVER_COOKIE_NOT_REQD (or
TFO_CLIENT_NO_COOKIE).
This is safe to do in certain environments where we know that there
isn't a malicous host (aka., data-centers) or when the
application-protocol already provides an authentication mechanism in the
first flight of data.

A server however might be providing multiple services or talking to both
sides (public Internet and data-center). So, this server would want to
enable cookie-less TFO for certain services and/or for connections that
go to the data-center.

This patch exposes a socket-option and a per-route attribute to enable such
fine-grained configurations.
Signed-off-by: NChristoph Paasch <cpaasch@apple.com>
Reviewed-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      71c02379
  13. 06 10月, 2017 1 次提交
    • E
      tcp: new list for sent but unacked skbs for RACK recovery · e2080072
      由 Eric Dumazet 提交于 
      This patch adds a new queue (list) that tracks the sent but not yet
acked or SACKed skbs for a TCP connection. The list is chronologically
ordered by skb->skb_mstamp (the head is the oldest sent skb).

This list will be used to optimize TCP Rack recovery, which checks
an skb's timestamp to judge if it has been lost and needs to be
retransmitted. Since TCP write queue is ordered by sequence instead
of sent time, RACK has to scan over the write queue to catch all
eligible packets to detect lost retransmission, and iterates through
SACKed skbs repeatedly.

Special cares for rare events:
1. TCP repair fakes skb transmission so the send queue needs adjusted
2. SACK reneging would require re-inserting SACKed skbs into the
   send queue. For now I believe it's not worth the complexity to
   make RACK work perfectly on SACK reneging, so we do nothing here.
3. Fast Open: currently for non-TFO, send-queue correctly queues
   the pure SYN packet. For TFO which queues a pure SYN and
   then a data packet, send-queue only queues the data packet but
   not the pure SYN due to the structure of TFO code. This is okay
   because the SYN receiver would never respond with a SACK on a
   missing SYN (i.e. SYN is never fast-retransmitted by SACK/RACK).

In order to not grow sk_buff, we use an union for the new list and
_skb_refdst/destructor fields. This is a bit complicated because
we need to make sure _skb_refdst and destructor are properly zeroed
before skb is cloned/copied at transmit, and before being freed.
Signed-off-by: NEric Dumazet <edumazet@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>
      e2080072
  15. 31 8月, 2017 1 次提交
  17. 07 8月, 2017 1 次提交
  19. 01 8月, 2017 2 次提交
    • F
      tcp: remove header prediction · 45f119bf
      由 Florian Westphal 提交于 
      Like prequeue, I am not sure this is overly useful nowadays.

If we receive a train of packets, GRO will aggregate them if the
headers are the same (HP predates GRO by several years) so we don't
get a per-packet benefit, only a per-aggregated-packet one.
Signed-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      45f119bf
    • F
      tcp: remove prequeue support · e7942d06
      由 Florian Westphal 提交于 
      prequeue is a tcp receive optimization that moves part of rx processing
from bh to process context.

This only works if the socket being processed belongs to a process that
is blocked in recv on that socket.

In practice, this doesn't happen anymore that often because nowadays
servers tend to use an event driven (epoll) model.

Even normal client applications (web browsers) commonly use many tcp
connections in parallel.

This has measureable impact only in netperf (which uses plain recv and
thus allows prequeue use) from host to locally running vm (~4%), however,
there were no changes when using netperf between two physical hosts with
ixgbe interfaces.
Signed-off-by: NFlorian Westphal <fw@strlen.de>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      e7942d06
  21. 18 5月, 2017 1 次提交
  23. 17 5月, 2017 1 次提交
    • E
      tcp: internal implementation for pacing · 218af599
      由 Eric Dumazet 提交于 
      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>
      218af599
  25. 27 4月, 2017 2 次提交
  27. 25 4月, 2017 1 次提交
    • W
      net/tcp_fastopen: Disable active side TFO in certain scenarios · cf1ef3f0
      由 Wei Wang 提交于 
      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>
      cf1ef3f0
  29. 04 2月, 2017 1 次提交
  31. 26 1月, 2017 1 次提交
    • W
      net/tcp-fastopen: Add new API support · 19f6d3f3
      由 Wei Wang 提交于 
      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>
      19f6d3f3
  33. 14 1月, 2017 6 次提交
  35. 06 12月, 2016 2 次提交
  37. 03 12月, 2016 1 次提交
    • F
      tcp: randomize tcp timestamp offsets for each connection · 95a22cae
      由 Florian Westphal 提交于 
      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>
      95a22cae
  39. 30 11月, 2016 2 次提交
    • F
      tcp: SOF_TIMESTAMPING_OPT_STATS option for SO_TIMESTAMPING · 1c885808
      由 Francis Yan 提交于 
      This patch exports the sender chronograph stats via the socket
SO_TIMESTAMPING channel. Currently we can instrument how long a
particular application unit of data was queued in TCP by tracking
SOF_TIMESTAMPING_TX_SOFTWARE and SOF_TIMESTAMPING_TX_SCHED. Having
these sender chronograph stats exported simultaneously along with
these timestamps allow further breaking down the various sender
limitation.  For example, a video server can tell if a particular
chunk of video on a connection takes a long time to deliver because
TCP was experiencing small receive window. It is not possible to
tell before this patch without packet traces.

To prepare these stats, the user needs to set
SOF_TIMESTAMPING_OPT_STATS and SOF_TIMESTAMPING_OPT_TSONLY flags
while requesting other SOF_TIMESTAMPING TX timestamps. When the
timestamps are available in the error queue, the stats are returned
in a separate control message of type SCM_TIMESTAMPING_OPT_STATS,
in a list of TLVs (struct nlattr) of types: TCP_NLA_BUSY_TIME,
TCP_NLA_RWND_LIMITED, TCP_NLA_SNDBUF_LIMITED. Unit is microsecond.
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>
      1c885808
    • F
      tcp: instrument tcp sender limits chronographs · 05b055e8
      由 Francis Yan 提交于 
      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>
      05b055e8
  41. 10 11月, 2016 1 次提交
  43. 21 9月, 2016 5 次提交
    • Y
      tcp: export data delivery rate · eb8329e0
      由 Yuchung Cheng 提交于 
      This commit export two new fields in struct tcp_info:

  tcpi_delivery_rate: The most recent goodput, as measured by
    tcp_rate_gen(). If the socket is limited by the sending
    application (e.g., no data to send), it reports the highest
    measurement instead of the most recent. The unit is bytes per
    second (like other rate fields in tcp_info).

  tcpi_delivery_rate_app_limited: A boolean indicating if the goodput
    was measured when the socket's throughput was limited by the
    sending application.

This delivery rate information can be useful for applications that
want to know the current throughput the TCP connection is seeing,
e.g. adaptive bitrate video streaming. It can also be very useful for
debugging or troubleshooting.
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>
      eb8329e0
    • S
      tcp: track application-limited rate samples · d7722e85
      由 Soheil Hassas Yeganeh 提交于 
      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>
      d7722e85
    • Y
      tcp: track data delivery rate for a TCP connection · b9f64820
      由 Yuchung Cheng 提交于 
      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>
      b9f64820
    • N
      tcp: count packets marked lost for a TCP connection · 0682e690
      由 Neal Cardwell 提交于 
      Count the number of packets that a TCP connection marks lost.

Congestion control modules can use this loss rate information for more
intelligent decisions about how fast to send.

Specifically, this is used in TCP BBR policer detection. BBR uses a
high packet loss rate as one signal in its policer detection and
policer bandwidth estimation algorithm.

The BBR policer detection algorithm cannot simply track retransmits,
because a retransmit can be (and often is) an indicator of packets
lost long, long ago. This is particularly true in a long CA_Loss
period that repairs the initial massive losses when a policer kicks
in.
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>
      0682e690
    • N
      tcp: use windowed min filter library for TCP min_rtt estimation · 64033892
      由 Neal Cardwell 提交于 
      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>
      64033892
  45. 09 9月, 2016 1 次提交
    • Y
      tcp: use an RB tree for ooo receive queue · 9f5afeae
      由 Yaogong Wang 提交于 
      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>
      9f5afeae
  47. 15 3月, 2016 1 次提交
    • M
      tcp: Add RFC4898 tcpEStatsPerfDataSegsOut/In · a44d6eac
      由 Martin KaFai Lau 提交于 
      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 a9d99ce2 ("tcp: fix tcpi_segs_in after connection establishment")
Signed-off-by: NMartin 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: NEric Dumazet <edumazet@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      a44d6eac
  49. 11 2月, 2016 1 次提交
  51. 08 2月, 2016 1 次提交
    • Y
      tcp: new delivery accounting · ddf1af6f
      由 Yuchung Cheng 提交于 
      This patch changes the accounting of how many packets are
newly acked or sacked when the sender receives an ACK.

The current approach basically computes

   newly_acked_sacked = (prior_packets - prior_sacked) -
                        (tp->packets_out - tp->sacked_out)

   where prior_packets and prior_sacked out are snapshot
   at the beginning of the ACK processing.

The new approach tracks the delivery information via a new
TCP state variable "delivered" which monotically increases
as new packets are delivered in order or out-of-order.

The reason for this change is that the current approach is
brittle that produces negative or inaccurate estimate.

   1) For non-SACK connections, an ACK that advances the SND.UNA
   could reset the DUPACK counters (tp->sacked_out) in
   tcp_process_loss() or tcp_fastretrans_alert(). This inflates
   the inflight suddenly and causes under-estimate or even
   negative estimate. Here is a real example:

                   before   after (processing ACK)
   packets_out     75       73
   sacked_out      23        0
   ca state        Loss     Open

   The old approach computes (75-23) - (73 - 0) = -21 delivered
   while the new approach computes 1 delivered since it
   considers the 2nd-24th packets are delivered OOO.

   2) MSS change would re-count packets_out and sacked_out so
   the estimate is in-accurate and can even become negative.
   E.g., the inflight is doubled when MSS is halved.

   3) Spurious retransmission signaled by DSACK is not accounted

The new approach is simpler and more robust. For SACK connections,
tp->delivered increments as packets are being acked or sacked in
SACK and ACK processing.

For non-sack connections, it's done in tcp_remove_reno_sacks() and
tcp_add_reno_sack(). When an ACK advances the SND.UNA, tp->delivered
is incremented by the number of packets ACKed (less the current
number of DUPACKs received plus one packet hole).  Upon receiving
a DUPACK, tp->delivered is incremented assuming one out-of-order
packet is delivered.

Upon receiving a DSACK, tp->delivered is incremtened assuming one
retransmission is delivered in tcp_sacktag_write_queue().
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NEric Dumazet <ncardwell@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      ddf1af6f