2005-04-17 06:20:36 +08:00
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/*
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* INET An implementation of the TCP/IP protocol suite for the LINUX
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* operating system. INET is implemented using the BSD Socket
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* interface as the means of communication with the user level.
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*
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* Definitions for the TCP module.
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*
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* Version: @(#)tcp.h 1.0.5 05/23/93
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*
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2005-05-06 07:16:16 +08:00
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* Authors: Ross Biro
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2005-04-17 06:20:36 +08:00
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* Fred N. van Kempen, <waltje@uWalt.NL.Mugnet.ORG>
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version
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* 2 of the License, or (at your option) any later version.
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*/
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#ifndef _TCP_H
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#define _TCP_H
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#define FASTRETRANS_DEBUG 1
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#include <linux/list.h>
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#include <linux/tcp.h>
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2011-11-24 09:12:59 +08:00
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#include <linux/bug.h>
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2005-04-17 06:20:36 +08:00
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#include <linux/slab.h>
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#include <linux/cache.h>
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#include <linux/percpu.h>
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2005-11-11 05:01:24 +08:00
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#include <linux/skbuff.h>
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2008-02-08 13:49:26 +08:00
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#include <linux/cryptohash.h>
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TCPCT part 1d: define TCP cookie option, extend existing struct's
Data structures are carefully composed to require minimal additions.
For example, the struct tcp_options_received cookie_plus variable fits
between existing 16-bit and 8-bit variables, requiring no additional
space (taking alignment into consideration). There are no additions to
tcp_request_sock, and only 1 pointer in tcp_sock.
This is a significantly revised implementation of an earlier (year-old)
patch that no longer applies cleanly, with permission of the original
author (Adam Langley):
http://thread.gmane.org/gmane.linux.network/102586
The principle difference is using a TCP option to carry the cookie nonce,
instead of a user configured offset in the data. This is more flexible and
less subject to user configuration error. Such a cookie option has been
suggested for many years, and is also useful without SYN data, allowing
several related concepts to use the same extension option.
"Re: SYN floods (was: does history repeat itself?)", September 9, 1996.
http://www.merit.net/mail.archives/nanog/1996-09/msg00235.html
"Re: what a new TCP header might look like", May 12, 1998.
ftp://ftp.isi.edu/end2end/end2end-interest-1998.mail
These functions will also be used in subsequent patches that implement
additional features.
Requires:
TCPCT part 1a: add request_values parameter for sending SYNACK
TCPCT part 1b: generate Responder Cookie secret
TCPCT part 1c: sysctl_tcp_cookie_size, socket option TCP_COOKIE_TRANSACTIONS
Signed-off-by: William.Allen.Simpson@gmail.com
Signed-off-by: David S. Miller <davem@davemloft.net>
2009-12-03 02:17:05 +08:00
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#include <linux/kref.h>
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2014-02-27 06:02:48 +08:00
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#include <linux/ktime.h>
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2005-08-10 11:11:08 +08:00
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#include <net/inet_connection_sock.h>
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2005-08-10 11:44:40 +08:00
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#include <net/inet_timewait_sock.h>
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2005-08-10 11:00:51 +08:00
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#include <net/inet_hashtables.h>
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2005-04-17 06:20:36 +08:00
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#include <net/checksum.h>
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[NET] Generalise TCP's struct open_request minisock infrastructure
Kept this first changeset minimal, without changing existing names to
ease peer review.
Basicaly tcp_openreq_alloc now receives the or_calltable, that in turn
has two new members:
->slab, that replaces tcp_openreq_cachep
->obj_size, to inform the size of the openreq descendant for
a specific protocol
The protocol specific fields in struct open_request were moved to a
class hierarchy, with the things that are common to all connection
oriented PF_INET protocols in struct inet_request_sock, the TCP ones
in tcp_request_sock, that is an inet_request_sock, that is an
open_request.
I.e. this uses the same approach used for the struct sock class
hierarchy, with sk_prot indicating if the protocol wants to use the
open_request infrastructure by filling in sk_prot->rsk_prot with an
or_calltable.
Results? Performance is improved and TCP v4 now uses only 64 bytes per
open request minisock, down from 96 without this patch :-)
Next changeset will rename some of the structs, fields and functions
mentioned above, struct or_calltable is way unclear, better name it
struct request_sock_ops, s/struct open_request/struct request_sock/g,
etc.
Signed-off-by: Arnaldo Carvalho de Melo <acme@ghostprotocols.net>
Signed-off-by: David S. Miller <davem@davemloft.net>
2005-06-19 13:46:52 +08:00
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#include <net/request_sock.h>
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2005-04-17 06:20:36 +08:00
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#include <net/sock.h>
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#include <net/snmp.h>
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#include <net/ip.h>
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2005-08-10 11:08:28 +08:00
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#include <net/tcp_states.h>
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2007-05-27 17:04:16 +08:00
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#include <net/inet_ecn.h>
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2009-05-05 02:11:01 +08:00
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#include <net/dst.h>
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2005-08-10 11:08:28 +08:00
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2005-04-17 06:20:36 +08:00
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#include <linux/seq_file.h>
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2011-12-12 05:47:02 +08:00
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#include <linux/memcontrol.h>
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2005-04-17 06:20:36 +08:00
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2005-08-10 11:07:35 +08:00
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extern struct inet_hashinfo tcp_hashinfo;
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2005-04-17 06:20:36 +08:00
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2008-11-26 13:17:14 +08:00
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extern struct percpu_counter tcp_orphan_count;
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2013-09-24 02:33:32 +08:00
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void tcp_time_wait(struct sock *sk, int state, int timeo);
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2005-04-17 06:20:36 +08:00
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#define MAX_TCP_HEADER (128 + MAX_HEADER)
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2008-07-19 15:04:31 +08:00
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#define MAX_TCP_OPTION_SPACE 40
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2005-04-17 06:20:36 +08:00
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2014-10-20 17:15:50 +08:00
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/*
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2005-04-17 06:20:36 +08:00
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* Never offer a window over 32767 without using window scaling. Some
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2014-10-20 17:15:50 +08:00
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* poor stacks do signed 16bit maths!
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2005-04-17 06:20:36 +08:00
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*/
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#define MAX_TCP_WINDOW 32767U
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/* Minimal accepted MSS. It is (60+60+8) - (20+20). */
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#define TCP_MIN_MSS 88U
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2006-03-21 09:53:41 +08:00
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/* The least MTU to use for probing */
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2015-03-06 11:18:22 +08:00
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#define TCP_BASE_MSS 1024
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2006-03-21 09:53:41 +08:00
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2015-03-06 11:18:24 +08:00
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/* probing interval, default to 10 minutes as per RFC4821 */
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#define TCP_PROBE_INTERVAL 600
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2015-03-06 11:18:23 +08:00
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/* Specify interval when tcp mtu probing will stop */
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#define TCP_PROBE_THRESHOLD 8
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2005-04-17 06:20:36 +08:00
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/* After receiving this amount of duplicate ACKs fast retransmit starts. */
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#define TCP_FASTRETRANS_THRESH 3
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/* Maximal number of ACKs sent quickly to accelerate slow-start. */
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#define TCP_MAX_QUICKACKS 16U
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2017-04-04 21:09:48 +08:00
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/* Maximal number of window scale according to RFC1323 */
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#define TCP_MAX_WSCALE 14U
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2005-04-17 06:20:36 +08:00
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/* urg_data states */
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#define TCP_URG_VALID 0x0100
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#define TCP_URG_NOTYET 0x0200
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#define TCP_URG_READ 0x0400
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#define TCP_RETR1 3 /*
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* This is how many retries it does before it
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* tries to figure out if the gateway is
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* down. Minimal RFC value is 3; it corresponds
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* to ~3sec-8min depending on RTO.
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*/
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#define TCP_RETR2 15 /*
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* This should take at least
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* 90 minutes to time out.
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* RFC1122 says that the limit is 100 sec.
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* 15 is ~13-30min depending on RTO.
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*/
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2012-08-31 10:48:31 +08:00
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#define TCP_SYN_RETRIES 6 /* This is how many retries are done
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* when active opening a connection.
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* RFC1122 says the minimum retry MUST
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* be at least 180secs. Nevertheless
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* this value is corresponding to
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* 63secs of retransmission with the
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* current initial RTO.
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*/
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2005-04-17 06:20:36 +08:00
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2012-08-31 10:48:31 +08:00
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#define TCP_SYNACK_RETRIES 5 /* This is how may retries are done
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* when passive opening a connection.
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* This is corresponding to 31secs of
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* retransmission with the current
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* initial RTO.
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*/
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2005-04-17 06:20:36 +08:00
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#define TCP_TIMEWAIT_LEN (60*HZ) /* how long to wait to destroy TIME-WAIT
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* state, about 60 seconds */
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#define TCP_FIN_TIMEOUT TCP_TIMEWAIT_LEN
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/* BSD style FIN_WAIT2 deadlock breaker.
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* It used to be 3min, new value is 60sec,
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* to combine FIN-WAIT-2 timeout with
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* TIME-WAIT timer.
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*/
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#define TCP_DELACK_MAX ((unsigned)(HZ/5)) /* maximal time to delay before sending an ACK */
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#if HZ >= 100
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#define TCP_DELACK_MIN ((unsigned)(HZ/25)) /* minimal time to delay before sending an ACK */
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#define TCP_ATO_MIN ((unsigned)(HZ/25))
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#else
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#define TCP_DELACK_MIN 4U
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#define TCP_ATO_MIN 4U
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#endif
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#define TCP_RTO_MAX ((unsigned)(120*HZ))
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#define TCP_RTO_MIN ((unsigned)(HZ/5))
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2012-04-13 03:48:40 +08:00
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#define TCP_TIMEOUT_INIT ((unsigned)(1*HZ)) /* RFC6298 2.1 initial RTO value */
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2011-06-08 19:08:38 +08:00
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#define TCP_TIMEOUT_FALLBACK ((unsigned)(3*HZ)) /* RFC 1122 initial RTO value, now
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* used as a fallback RTO for the
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* initial data transmission if no
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* valid RTT sample has been acquired,
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* most likely due to retrans in 3WHS.
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*/
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2005-04-17 06:20:36 +08:00
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#define TCP_RESOURCE_PROBE_INTERVAL ((unsigned)(HZ/2U)) /* Maximal interval between probes
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* for local resources.
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*/
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2017-01-13 14:11:33 +08:00
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#define TCP_REO_TIMEOUT_MIN (2000) /* Min RACK reordering timeout in usec */
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2005-04-17 06:20:36 +08:00
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#define TCP_KEEPALIVE_TIME (120*60*HZ) /* two hours */
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#define TCP_KEEPALIVE_PROBES 9 /* Max of 9 keepalive probes */
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#define TCP_KEEPALIVE_INTVL (75*HZ)
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#define MAX_TCP_KEEPIDLE 32767
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#define MAX_TCP_KEEPINTVL 32767
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#define MAX_TCP_KEEPCNT 127
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#define MAX_TCP_SYNCNT 127
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#define TCP_SYNQ_INTERVAL (HZ/5) /* Period of SYNACK timer */
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#define TCP_PAWS_24DAYS (60 * 60 * 24 * 24)
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#define TCP_PAWS_MSL 60 /* Per-host timestamps are invalidated
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* after this time. It should be equal
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* (or greater than) TCP_TIMEWAIT_LEN
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* to provide reliability equal to one
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* provided by timewait state.
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*/
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#define TCP_PAWS_WINDOW 1 /* Replay window for per-host
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* timestamps. It must be less than
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* minimal timewait lifetime.
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*/
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/*
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* TCP option
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*/
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2014-10-20 17:15:50 +08:00
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2005-04-17 06:20:36 +08:00
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#define TCPOPT_NOP 1 /* Padding */
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#define TCPOPT_EOL 0 /* End of options */
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#define TCPOPT_MSS 2 /* Segment size negotiating */
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#define TCPOPT_WINDOW 3 /* Window scaling */
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#define TCPOPT_SACK_PERM 4 /* SACK Permitted */
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#define TCPOPT_SACK 5 /* SACK Block */
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#define TCPOPT_TIMESTAMP 8 /* Better RTT estimations/PAWS */
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2006-11-15 11:07:45 +08:00
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#define TCPOPT_MD5SIG 19 /* MD5 Signature (RFC2385) */
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2015-04-07 05:37:26 +08:00
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#define TCPOPT_FASTOPEN 34 /* Fast open (RFC7413) */
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2012-07-19 14:43:05 +08:00
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#define TCPOPT_EXP 254 /* Experimental */
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/* Magic number to be after the option value for sharing TCP
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* experimental options. See draft-ietf-tcpm-experimental-options-00.txt
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*/
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#define TCPOPT_FASTOPEN_MAGIC 0xF989
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2005-04-17 06:20:36 +08:00
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/*
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* TCP option lengths
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*/
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#define TCPOLEN_MSS 4
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#define TCPOLEN_WINDOW 3
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#define TCPOLEN_SACK_PERM 2
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#define TCPOLEN_TIMESTAMP 10
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2006-11-15 11:07:45 +08:00
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#define TCPOLEN_MD5SIG 18
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2015-04-07 05:37:26 +08:00
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#define TCPOLEN_FASTOPEN_BASE 2
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2012-07-19 14:43:05 +08:00
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#define TCPOLEN_EXP_FASTOPEN_BASE 4
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2005-04-17 06:20:36 +08:00
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/* But this is what stacks really send out. */
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#define TCPOLEN_TSTAMP_ALIGNED 12
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#define TCPOLEN_WSCALE_ALIGNED 4
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#define TCPOLEN_SACKPERM_ALIGNED 4
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#define TCPOLEN_SACK_BASE 2
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#define TCPOLEN_SACK_BASE_ALIGNED 4
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#define TCPOLEN_SACK_PERBLOCK 8
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2006-11-15 11:07:45 +08:00
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#define TCPOLEN_MD5SIG_ALIGNED 20
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2008-07-19 15:04:31 +08:00
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#define TCPOLEN_MSS_ALIGNED 4
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2005-04-17 06:20:36 +08:00
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/* Flags in tp->nonagle */
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#define TCP_NAGLE_OFF 1 /* Nagle's algo is disabled */
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#define TCP_NAGLE_CORK 2 /* Socket is corked */
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2005-11-11 09:13:47 +08:00
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#define TCP_NAGLE_PUSH 4 /* Cork is overridden for already queued data */
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2005-04-17 06:20:36 +08:00
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2010-02-18 10:47:01 +08:00
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/* TCP thin-stream limits */
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#define TCP_THIN_LINEAR_RETRIES 6 /* After 6 linear retries, do exp. backoff */
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2016-01-28 05:54:06 +08:00
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/* TCP initial congestion window as per rfc6928 */
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2011-02-03 09:05:11 +08:00
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#define TCP_INIT_CWND 10
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2012-07-19 14:43:09 +08:00
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/* Bit Flags for sysctl_tcp_fastopen */
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#define TFO_CLIENT_ENABLE 1
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2012-08-31 20:29:11 +08:00
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#define TFO_SERVER_ENABLE 2
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2012-07-19 14:43:11 +08:00
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#define TFO_CLIENT_NO_COOKIE 4 /* Data in SYN w/o cookie option */
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2012-07-19 14:43:09 +08:00
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2012-08-31 20:29:11 +08:00
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/* Accept SYN data w/o any cookie option */
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#define TFO_SERVER_COOKIE_NOT_REQD 0x200
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/* Force enable TFO on all listeners, i.e., not requiring the
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2016-08-23 08:17:54 +08:00
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* TCP_FASTOPEN socket option.
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2012-08-31 20:29:11 +08:00
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*/
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#define TFO_SERVER_WO_SOCKOPT1 0x400
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2005-08-10 11:44:40 +08:00
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2005-04-17 06:20:36 +08:00
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/* sysctl variables for tcp */
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extern int sysctl_tcp_timestamps;
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extern int sysctl_tcp_window_scaling;
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extern int sysctl_tcp_sack;
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2012-07-19 14:43:05 +08:00
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extern int sysctl_tcp_fastopen;
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2005-04-17 06:20:36 +08:00
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extern int sysctl_tcp_retrans_collapse;
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extern int sysctl_tcp_stdurg;
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extern int sysctl_tcp_rfc1337;
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extern int sysctl_tcp_abort_on_overflow;
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extern int sysctl_tcp_max_orphans;
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extern int sysctl_tcp_fack;
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extern int sysctl_tcp_reordering;
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2014-10-28 12:45:24 +08:00
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extern int sysctl_tcp_max_reordering;
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2005-04-17 06:20:36 +08:00
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extern int sysctl_tcp_dsack;
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2013-10-20 07:25:36 +08:00
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extern long sysctl_tcp_mem[3];
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2005-04-17 06:20:36 +08:00
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extern int sysctl_tcp_wmem[3];
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extern int sysctl_tcp_rmem[3];
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extern int sysctl_tcp_app_win;
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extern int sysctl_tcp_adv_win_scale;
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extern int sysctl_tcp_frto;
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extern int sysctl_tcp_low_latency;
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extern int sysctl_tcp_nometrics_save;
|
|
|
|
extern int sysctl_tcp_moderate_rcvbuf;
|
|
|
|
extern int sysctl_tcp_tso_win_divisor;
|
2006-03-21 14:40:29 +08:00
|
|
|
extern int sysctl_tcp_workaround_signed_windows;
|
2006-06-14 13:33:04 +08:00
|
|
|
extern int sysctl_tcp_slow_start_after_idle;
|
2010-02-18 10:47:01 +08:00
|
|
|
extern int sysctl_tcp_thin_linear_timeouts;
|
2010-02-18 12:48:19 +08:00
|
|
|
extern int sysctl_tcp_thin_dupack;
|
2012-05-02 21:30:03 +08:00
|
|
|
extern int sysctl_tcp_early_retrans;
|
2017-01-13 14:11:36 +08:00
|
|
|
extern int sysctl_tcp_recovery;
|
|
|
|
#define TCP_RACK_LOSS_DETECTION 0x1 /* Use RACK to detect losses */
|
|
|
|
|
tcp: TCP Small Queues
This introduce TSQ (TCP Small Queues)
TSQ goal is to reduce number of TCP packets in xmit queues (qdisc &
device queues), to reduce RTT and cwnd bias, part of the bufferbloat
problem.
sk->sk_wmem_alloc not allowed to grow above a given limit,
allowing no more than ~128KB [1] per tcp socket in qdisc/dev layers at a
given time.
TSO packets are sized/capped to half the limit, so that we have two
TSO packets in flight, allowing better bandwidth use.
As a side effect, setting the limit to 40000 automatically reduces the
standard gso max limit (65536) to 40000/2 : It can help to reduce
latencies of high prio packets, having smaller TSO packets.
This means we divert sock_wfree() to a tcp_wfree() handler, to
queue/send following frames when skb_orphan() [2] is called for the
already queued skbs.
Results on my dev machines (tg3/ixgbe nics) are really impressive,
using standard pfifo_fast, and with or without TSO/GSO.
Without reduction of nominal bandwidth, we have reduction of buffering
per bulk sender :
< 1ms on Gbit (instead of 50ms with TSO)
< 8ms on 100Mbit (instead of 132 ms)
I no longer have 4 MBytes backlogged in qdisc by a single netperf
session, and both side socket autotuning no longer use 4 Mbytes.
As skb destructor cannot restart xmit itself ( as qdisc lock might be
taken at this point ), we delegate the work to a tasklet. We use one
tasklest per cpu for performance reasons.
If tasklet finds a socket owned by the user, it sets TSQ_OWNED flag.
This flag is tested in a new protocol method called from release_sock(),
to eventually send new segments.
[1] New /proc/sys/net/ipv4/tcp_limit_output_bytes tunable
[2] skb_orphan() is usually called at TX completion time,
but some drivers call it in their start_xmit() handler.
These drivers should at least use BQL, or else a single TCP
session can still fill the whole NIC TX ring, since TSQ will
have no effect.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Dave Taht <dave.taht@bufferbloat.net>
Cc: Tom Herbert <therbert@google.com>
Cc: Matt Mathis <mattmathis@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2012-07-11 13:50:31 +08:00
|
|
|
extern int sysctl_tcp_limit_output_bytes;
|
2012-07-17 16:13:05 +08:00
|
|
|
extern int sysctl_tcp_challenge_ack_limit;
|
tcp: TSO packets automatic sizing
After hearing many people over past years complaining against TSO being
bursty or even buggy, we are proud to present automatic sizing of TSO
packets.
One part of the problem is that tcp_tso_should_defer() uses an heuristic
relying on upcoming ACKS instead of a timer, but more generally, having
big TSO packets makes little sense for low rates, as it tends to create
micro bursts on the network, and general consensus is to reduce the
buffering amount.
This patch introduces a per socket sk_pacing_rate, that approximates
the current sending rate, and allows us to size the TSO packets so
that we try to send one packet every ms.
This field could be set by other transports.
Patch has no impact for high speed flows, where having large TSO packets
makes sense to reach line rate.
For other flows, this helps better packet scheduling and ACK clocking.
This patch increases performance of TCP flows in lossy environments.
A new sysctl (tcp_min_tso_segs) is added, to specify the
minimal size of a TSO packet (default being 2).
A follow-up patch will provide a new packet scheduler (FQ), using
sk_pacing_rate as an input to perform optional per flow pacing.
This explains why we chose to set sk_pacing_rate to twice the current
rate, allowing 'slow start' ramp up.
sk_pacing_rate = 2 * cwnd * mss / srtt
v2: Neal Cardwell reported a suspect deferring of last two segments on
initial write of 10 MSS, I had to change tcp_tso_should_defer() to take
into account tp->xmit_size_goal_segs
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Van Jacobson <vanj@google.com>
Cc: Tom Herbert <therbert@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2013-08-27 20:46:32 +08:00
|
|
|
extern int sysctl_tcp_min_tso_segs;
|
2015-10-17 12:57:42 +08:00
|
|
|
extern int sysctl_tcp_min_rtt_wlen;
|
tcp: auto corking
With the introduction of TCP Small Queues, TSO auto sizing, and TCP
pacing, we can implement Automatic Corking in the kernel, to help
applications doing small write()/sendmsg() to TCP sockets.
Idea is to change tcp_push() to check if the current skb payload is
under skb optimal size (a multiple of MSS bytes)
If under 'size_goal', and at least one packet is still in Qdisc or
NIC TX queues, set the TCP Small Queue Throttled bit, so that the push
will be delayed up to TX completion time.
This delay might allow the application to coalesce more bytes
in the skb in following write()/sendmsg()/sendfile() system calls.
The exact duration of the delay is depending on the dynamics
of the system, and might be zero if no packet for this flow
is actually held in Qdisc or NIC TX ring.
Using FQ/pacing is a way to increase the probability of
autocorking being triggered.
Add a new sysctl (/proc/sys/net/ipv4/tcp_autocorking) to control
this feature and default it to 1 (enabled)
Add a new SNMP counter : nstat -a | grep TcpExtTCPAutoCorking
This counter is incremented every time we detected skb was under used
and its flush was deferred.
Tested:
Interesting effects when using line buffered commands under ssh.
Excellent performance results in term of cpu usage and total throughput.
lpq83:~# echo 1 >/proc/sys/net/ipv4/tcp_autocorking
lpq83:~# perf stat ./super_netperf 4 -t TCP_STREAM -H lpq84 -- -m 128
9410.39
Performance counter stats for './super_netperf 4 -t TCP_STREAM -H lpq84 -- -m 128':
35209.439626 task-clock # 2.901 CPUs utilized
2,294 context-switches # 0.065 K/sec
101 CPU-migrations # 0.003 K/sec
4,079 page-faults # 0.116 K/sec
97,923,241,298 cycles # 2.781 GHz [83.31%]
51,832,908,236 stalled-cycles-frontend # 52.93% frontend cycles idle [83.30%]
25,697,986,603 stalled-cycles-backend # 26.24% backend cycles idle [66.70%]
102,225,978,536 instructions # 1.04 insns per cycle
# 0.51 stalled cycles per insn [83.38%]
18,657,696,819 branches # 529.906 M/sec [83.29%]
91,679,646 branch-misses # 0.49% of all branches [83.40%]
12.136204899 seconds time elapsed
lpq83:~# echo 0 >/proc/sys/net/ipv4/tcp_autocorking
lpq83:~# perf stat ./super_netperf 4 -t TCP_STREAM -H lpq84 -- -m 128
6624.89
Performance counter stats for './super_netperf 4 -t TCP_STREAM -H lpq84 -- -m 128':
40045.864494 task-clock # 3.301 CPUs utilized
171 context-switches # 0.004 K/sec
53 CPU-migrations # 0.001 K/sec
4,080 page-faults # 0.102 K/sec
111,340,458,645 cycles # 2.780 GHz [83.34%]
61,778,039,277 stalled-cycles-frontend # 55.49% frontend cycles idle [83.31%]
29,295,522,759 stalled-cycles-backend # 26.31% backend cycles idle [66.67%]
108,654,349,355 instructions # 0.98 insns per cycle
# 0.57 stalled cycles per insn [83.34%]
19,552,170,748 branches # 488.244 M/sec [83.34%]
157,875,417 branch-misses # 0.81% of all branches [83.34%]
12.130267788 seconds time elapsed
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2013-12-06 14:36:05 +08:00
|
|
|
extern int sysctl_tcp_autocorking;
|
tcp: helpers to mitigate ACK loops by rate-limiting out-of-window dupacks
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>
2015-02-07 05:04:38 +08:00
|
|
|
extern int sysctl_tcp_invalid_ratelimit;
|
2015-08-22 08:38:02 +08:00
|
|
|
extern int sysctl_tcp_pacing_ss_ratio;
|
|
|
|
extern int sysctl_tcp_pacing_ca_ratio;
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2010-11-10 07:24:26 +08:00
|
|
|
extern atomic_long_t tcp_memory_allocated;
|
2008-11-26 13:16:35 +08:00
|
|
|
extern struct percpu_counter tcp_sockets_allocated;
|
2005-04-17 06:20:36 +08:00
|
|
|
extern int tcp_memory_pressure;
|
|
|
|
|
2015-05-16 03:39:27 +08:00
|
|
|
/* optimized version of sk_under_memory_pressure() for TCP sockets */
|
|
|
|
static inline bool tcp_under_memory_pressure(const struct sock *sk)
|
|
|
|
{
|
2016-01-15 07:21:17 +08:00
|
|
|
if (mem_cgroup_sockets_enabled && sk->sk_memcg &&
|
|
|
|
mem_cgroup_under_socket_pressure(sk->sk_memcg))
|
net: tcp_memcontrol: sanitize tcp memory accounting callbacks
There won't be a tcp control soft limit, so integrating the memcg code
into the global skmem limiting scheme complicates things unnecessarily.
Replace this with simple and clear charge and uncharge calls--hidden
behind a jump label--to account skb memory.
Note that this is not purely aesthetic: as a result of shoehorning the
per-memcg code into the same memory accounting functions that handle the
global level, the old code would compare the per-memcg consumption
against the smaller of the per-memcg limit and the global limit. This
allowed the total consumption of multiple sockets to exceed the global
limit, as long as the individual sockets stayed within bounds. After
this change, the code will always compare the per-memcg consumption to
the per-memcg limit, and the global consumption to the global limit, and
thus close this loophole.
Without a soft limit, the per-memcg memory pressure state in sockets is
generally questionable. However, we did it until now, so we continue to
enter it when the hard limit is hit, and packets are dropped, to let
other sockets in the cgroup know that they shouldn't grow their transmit
windows, either. However, keep it simple in the new callback model and
leave memory pressure lazily when the next packet is accepted (as
opposed to doing it synchroneously when packets are processed). When
packets are dropped, network performance will already be in the toilet,
so that should be a reasonable trade-off.
As described above, consumption is now checked on the per-memcg level
and the global level separately. Likewise, memory pressure states are
maintained on both the per-memcg level and the global level, and a
socket is considered under pressure when either level asserts as much.
Signed-off-by: Johannes Weiner <hannes@cmpxchg.org>
Reviewed-by: Vladimir Davydov <vdavydov@virtuozzo.com>
Acked-by: David S. Miller <davem@davemloft.net>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2016-01-15 07:21:14 +08:00
|
|
|
return true;
|
2015-05-16 03:39:27 +08:00
|
|
|
|
|
|
|
return tcp_memory_pressure;
|
|
|
|
}
|
2005-04-17 06:20:36 +08:00
|
|
|
/*
|
|
|
|
* The next routines deal with comparing 32 bit unsigned ints
|
|
|
|
* and worry about wraparound (automatic with unsigned arithmetic).
|
|
|
|
*/
|
|
|
|
|
2012-05-17 07:15:34 +08:00
|
|
|
static inline bool before(__u32 seq1, __u32 seq2)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
2007-01-05 04:25:16 +08:00
|
|
|
return (__s32)(seq1-seq2) < 0;
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
[TCP]: Fix ambiguity in the `before' relation.
While looking at DCCP sequence numbers, I stumbled over a problem with
the following definition of before in tcp.h:
static inline int before(__u32 seq1, __u32 seq2)
{
return (__s32)(seq1-seq2) < 0;
}
Problem: This definition suffers from an an ambiguity, i.e. always
before(a, (a + 2^31) % 2^32)) = 1
before((a + 2^31) % 2^32), a) = 1
In text: when the difference between a and b amounts to 2^31,
a is always considered `before' b, the function can not decide.
The reason is that implicitly 0 is `before' 1 ... 2^31-1 ... 2^31
Solution: There is a simple fix, by defining before in such a way that
0 is no longer `before' 2^31, i.e. 0 `before' 1 ... 2^31-1
By not using the middle between 0 and 2^32, before can be made
unambiguous.
This is achieved by testing whether seq2-seq1 > 0 (using signed
32-bit arithmetic).
I attach a patch to codify this. Also the `after' relation is basically
a redefinition of `before', it is now defined as a macro after before.
Signed-off-by: Gerrit Renker <gerrit@erg.abdn.ac.uk>
Signed-off-by: David S. Miller <davem@davemloft.net>
2006-12-21 02:25:55 +08:00
|
|
|
#define after(seq2, seq1) before(seq1, seq2)
|
2005-04-17 06:20:36 +08:00
|
|
|
|
|
|
|
/* is s2<=s1<=s3 ? */
|
2012-05-17 07:15:34 +08:00
|
|
|
static inline bool between(__u32 seq1, __u32 seq2, __u32 seq3)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
|
|
|
return seq3 - seq2 >= seq1 - seq2;
|
|
|
|
}
|
|
|
|
|
2012-01-31 06:16:06 +08:00
|
|
|
static inline bool tcp_out_of_memory(struct sock *sk)
|
|
|
|
{
|
|
|
|
if (sk->sk_wmem_queued > SOCK_MIN_SNDBUF &&
|
|
|
|
sk_memory_allocated(sk) > sk_prot_mem_limits(sk, 2))
|
|
|
|
return true;
|
|
|
|
return false;
|
|
|
|
}
|
|
|
|
|
2015-05-16 03:39:26 +08:00
|
|
|
void sk_forced_mem_schedule(struct sock *sk, int size);
|
|
|
|
|
2010-08-25 17:27:49 +08:00
|
|
|
static inline bool tcp_too_many_orphans(struct sock *sk, int shift)
|
2007-05-30 04:19:18 +08:00
|
|
|
{
|
2010-08-25 17:27:49 +08:00
|
|
|
struct percpu_counter *ocp = sk->sk_prot->orphan_count;
|
|
|
|
int orphans = percpu_counter_read_positive(ocp);
|
|
|
|
|
|
|
|
if (orphans << shift > sysctl_tcp_max_orphans) {
|
|
|
|
orphans = percpu_counter_sum_positive(ocp);
|
|
|
|
if (orphans << shift > sysctl_tcp_max_orphans)
|
|
|
|
return true;
|
|
|
|
}
|
|
|
|
return false;
|
2007-05-30 04:19:18 +08:00
|
|
|
}
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
bool tcp_check_oom(struct sock *sk, int shift);
|
2012-01-31 06:16:06 +08:00
|
|
|
|
2009-04-19 17:43:48 +08:00
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
extern struct proto tcp_prot;
|
|
|
|
|
2008-07-18 19:02:08 +08:00
|
|
|
#define TCP_INC_STATS(net, field) SNMP_INC_STATS((net)->mib.tcp_statistics, field)
|
2016-04-28 07:44:43 +08:00
|
|
|
#define __TCP_INC_STATS(net, field) __SNMP_INC_STATS((net)->mib.tcp_statistics, field)
|
2008-07-18 19:02:08 +08:00
|
|
|
#define TCP_DEC_STATS(net, field) SNMP_DEC_STATS((net)->mib.tcp_statistics, field)
|
2010-04-22 15:00:24 +08:00
|
|
|
#define TCP_ADD_STATS(net, field, val) SNMP_ADD_STATS((net)->mib.tcp_statistics, field, val)
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_tasklet_init(void);
|
|
|
|
|
|
|
|
void tcp_v4_err(struct sk_buff *skb, u32);
|
|
|
|
|
|
|
|
void tcp_shutdown(struct sock *sk, int how);
|
|
|
|
|
|
|
|
void tcp_v4_early_demux(struct sk_buff *skb);
|
|
|
|
int tcp_v4_rcv(struct sk_buff *skb);
|
|
|
|
|
|
|
|
int tcp_v4_tw_remember_stamp(struct inet_timewait_sock *tw);
|
2015-03-02 15:37:48 +08:00
|
|
|
int tcp_sendmsg(struct sock *sk, struct msghdr *msg, size_t size);
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_sendpage(struct sock *sk, struct page *page, int offset, size_t size,
|
|
|
|
int flags);
|
|
|
|
void tcp_release_cb(struct sock *sk);
|
|
|
|
void tcp_wfree(struct sk_buff *skb);
|
|
|
|
void tcp_write_timer_handler(struct sock *sk);
|
|
|
|
void tcp_delack_timer_handler(struct sock *sk);
|
|
|
|
int tcp_ioctl(struct sock *sk, int cmd, unsigned long arg);
|
2015-09-29 22:42:41 +08:00
|
|
|
int tcp_rcv_state_process(struct sock *sk, struct sk_buff *skb);
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_rcv_established(struct sock *sk, struct sk_buff *skb,
|
|
|
|
const struct tcphdr *th, unsigned int len);
|
|
|
|
void tcp_rcv_space_adjust(struct sock *sk);
|
|
|
|
int tcp_twsk_unique(struct sock *sk, struct sock *sktw, void *twp);
|
|
|
|
void tcp_twsk_destructor(struct sock *sk);
|
|
|
|
ssize_t tcp_splice_read(struct socket *sk, loff_t *ppos,
|
|
|
|
struct pipe_inode_info *pipe, size_t len,
|
|
|
|
unsigned int flags);
|
2007-11-07 15:30:13 +08:00
|
|
|
|
2005-08-10 11:10:42 +08:00
|
|
|
static inline void tcp_dec_quickack_mode(struct sock *sk,
|
|
|
|
const unsigned int pkts)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
2005-08-10 11:10:42 +08:00
|
|
|
struct inet_connection_sock *icsk = inet_csk(sk);
|
2005-07-06 06:17:45 +08:00
|
|
|
|
2005-08-10 11:10:42 +08:00
|
|
|
if (icsk->icsk_ack.quick) {
|
|
|
|
if (pkts >= icsk->icsk_ack.quick) {
|
|
|
|
icsk->icsk_ack.quick = 0;
|
2005-07-06 06:17:45 +08:00
|
|
|
/* Leaving quickack mode we deflate ATO. */
|
2005-08-10 11:10:42 +08:00
|
|
|
icsk->icsk_ack.ato = TCP_ATO_MIN;
|
2005-07-06 06:17:45 +08:00
|
|
|
} else
|
2005-08-10 11:10:42 +08:00
|
|
|
icsk->icsk_ack.quick -= pkts;
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2007-05-27 17:04:16 +08:00
|
|
|
#define TCP_ECN_OK 1
|
|
|
|
#define TCP_ECN_QUEUE_CWR 2
|
|
|
|
#define TCP_ECN_DEMAND_CWR 4
|
2011-09-23 04:02:19 +08:00
|
|
|
#define TCP_ECN_SEEN 8
|
2007-05-27 17:04:16 +08:00
|
|
|
|
2009-11-03 11:26:03 +08:00
|
|
|
enum tcp_tw_status {
|
2005-04-17 06:20:36 +08:00
|
|
|
TCP_TW_SUCCESS = 0,
|
|
|
|
TCP_TW_RST = 1,
|
|
|
|
TCP_TW_ACK = 2,
|
|
|
|
TCP_TW_SYN = 3
|
|
|
|
};
|
|
|
|
|
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
enum tcp_tw_status tcp_timewait_state_process(struct inet_timewait_sock *tw,
|
|
|
|
struct sk_buff *skb,
|
|
|
|
const struct tcphdr *th);
|
|
|
|
struct sock *tcp_check_req(struct sock *sk, struct sk_buff *skb,
|
2015-03-20 10:04:19 +08:00
|
|
|
struct request_sock *req, bool fastopen);
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_child_process(struct sock *parent, struct sock *child,
|
|
|
|
struct sk_buff *skb);
|
tcp: reduce spurious retransmits due to transient SACK reneging
This commit reduces spurious retransmits due to apparent SACK reneging
by only reacting to SACK reneging that persists for a short delay.
When a sequence space hole at snd_una is filled, some TCP receivers
send a series of ACKs as they apparently scan their out-of-order queue
and cumulatively ACK all the packets that have now been consecutiveyly
received. This is essentially misbehavior B in "Misbehaviors in TCP
SACK generation" ACM SIGCOMM Computer Communication Review, April
2011, so we suspect that this is from several common OSes (Windows
2000, Windows Server 2003, Windows XP). However, this issue has also
been seen in other cases, e.g. the netdev thread "TCP being hoodwinked
into spurious retransmissions by lack of timestamps?" from March 2014,
where the receiver was thought to be a BSD box.
Since snd_una would temporarily be adjacent to a previously SACKed
range in these scenarios, this receiver behavior triggered the Linux
SACK reneging code path in the sender. This led the sender to clear
the SACK scoreboard, enter CA_Loss, and spuriously retransmit
(potentially) every packet from the entire write queue at line rate
just a few milliseconds before the ACK for each packet arrives at the
sender.
To avoid such situations, now when a sender sees apparent reneging it
does not yet retransmit, but rather adjusts the RTO timer to give the
receiver a little time (max(RTT/2, 10ms)) to send us some more ACKs
that will restore sanity to the SACK scoreboard. If the reneging
persists until this RTO then, as before, we clear the SACK scoreboard
and enter CA_Loss.
A 10ms delay tolerates a receiver sending such a stream of ACKs at
56Kbit/sec. And to allow for receivers with slower or more congested
paths, we wait for at least RTT/2.
We validated the resulting max(RTT/2, 10ms) delay formula with a mix
of North American and South American Google web server traffic, and
found that for ACKs displaying transient reneging:
(1) 90% of inter-ACK delays were less than 10ms
(2) 99% of inter-ACK delays were less than RTT/2
In tests on Google web servers this commit reduced reneging events by
75%-90% (as measured by the TcpExtTCPSACKReneging counter), without
any measurable impact on latency for user HTTP and SPDY requests.
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2014-08-05 07:12:29 +08:00
|
|
|
void tcp_enter_loss(struct sock *sk);
|
2017-01-13 14:11:33 +08:00
|
|
|
void tcp_cwnd_reduction(struct sock *sk, int newly_acked_sacked, int flag);
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_clear_retrans(struct tcp_sock *tp);
|
|
|
|
void tcp_update_metrics(struct sock *sk);
|
|
|
|
void tcp_init_metrics(struct sock *sk);
|
|
|
|
void tcp_metrics_init(void);
|
2017-03-16 04:30:45 +08:00
|
|
|
bool tcp_peer_is_proven(struct request_sock *req, struct dst_entry *dst);
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_disable_fack(struct tcp_sock *tp);
|
|
|
|
void tcp_close(struct sock *sk, long timeout);
|
|
|
|
void tcp_init_sock(struct sock *sk);
|
|
|
|
unsigned int tcp_poll(struct file *file, struct socket *sock,
|
|
|
|
struct poll_table_struct *wait);
|
|
|
|
int tcp_getsockopt(struct sock *sk, int level, int optname,
|
|
|
|
char __user *optval, int __user *optlen);
|
|
|
|
int tcp_setsockopt(struct sock *sk, int level, int optname,
|
|
|
|
char __user *optval, unsigned int optlen);
|
|
|
|
int compat_tcp_getsockopt(struct sock *sk, int level, int optname,
|
2010-07-11 04:41:06 +08:00
|
|
|
char __user *optval, int __user *optlen);
|
2013-09-24 02:33:32 +08:00
|
|
|
int compat_tcp_setsockopt(struct sock *sk, int level, int optname,
|
2010-07-11 04:41:06 +08:00
|
|
|
char __user *optval, unsigned int optlen);
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_set_keepalive(struct sock *sk, int val);
|
2015-03-23 01:22:19 +08:00
|
|
|
void tcp_syn_ack_timeout(const struct request_sock *req);
|
2015-03-02 15:37:48 +08:00
|
|
|
int tcp_recvmsg(struct sock *sk, struct msghdr *msg, size_t len, int nonblock,
|
|
|
|
int flags, int *addr_len);
|
2017-06-08 01:34:36 +08:00
|
|
|
void tcp_parse_options(const struct net *net, const struct sk_buff *skb,
|
2013-09-24 02:33:32 +08:00
|
|
|
struct tcp_options_received *opt_rx,
|
|
|
|
int estab, struct tcp_fastopen_cookie *foc);
|
|
|
|
const u8 *tcp_parse_md5sig_option(const struct tcphdr *th);
|
2008-04-17 11:29:53 +08:00
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
/*
|
|
|
|
* TCP v4 functions exported for the inet6 API
|
|
|
|
*/
|
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_v4_send_check(struct sock *sk, struct sk_buff *skb);
|
2014-08-15 00:40:05 +08:00
|
|
|
void tcp_v4_mtu_reduced(struct sock *sk);
|
2016-02-03 11:31:12 +08:00
|
|
|
void tcp_req_err(struct sock *sk, u32 seq, bool abort);
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_v4_conn_request(struct sock *sk, struct sk_buff *skb);
|
2015-09-29 22:42:47 +08:00
|
|
|
struct sock *tcp_create_openreq_child(const struct sock *sk,
|
2013-09-24 02:33:32 +08:00
|
|
|
struct request_sock *req,
|
|
|
|
struct sk_buff *skb);
|
net: tcp: add per route congestion control
This work adds the possibility to define a per route/destination
congestion control algorithm. Generally, this opens up the possibility
for a machine with different links to enforce specific congestion
control algorithms with optimal strategies for each of them based
on their network characteristics, even transparently for a single
application listening on all links.
For our specific use case, this additionally facilitates deployment
of DCTCP, for example, applications can easily serve internal
traffic/dsts in DCTCP and external one with CUBIC. Other scenarios
would also allow for utilizing e.g. long living, low priority
background flows for certain destinations/routes while still being
able for normal traffic to utilize the default congestion control
algorithm. We also thought about a per netns setting (where different
defaults are possible), but given its actually a link specific
property, we argue that a per route/destination setting is the most
natural and flexible.
The administrator can utilize this through ip-route(8) by appending
"congctl [lock] <name>", where <name> denotes the name of a
congestion control algorithm and the optional lock parameter allows
to enforce the given algorithm so that applications in user space
would not be allowed to overwrite that algorithm for that destination.
The dst metric lookups are being done when a dst entry is already
available in order to avoid a costly lookup and still before the
algorithms are being initialized, thus overhead is very low when the
feature is not being used. While the client side would need to drop
the current reference on the module, on server side this can actually
even be avoided as we just got a flat-copied socket clone.
Joint work with Florian Westphal.
Suggested-by: Hannes Frederic Sowa <hannes@stressinduktion.org>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:48 +08:00
|
|
|
void tcp_ca_openreq_child(struct sock *sk, const struct dst_entry *dst);
|
2015-09-29 22:42:48 +08:00
|
|
|
struct sock *tcp_v4_syn_recv_sock(const struct sock *sk, struct sk_buff *skb,
|
2013-09-24 02:33:32 +08:00
|
|
|
struct request_sock *req,
|
2015-10-22 23:20:46 +08:00
|
|
|
struct dst_entry *dst,
|
|
|
|
struct request_sock *req_unhash,
|
|
|
|
bool *own_req);
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_v4_do_rcv(struct sock *sk, struct sk_buff *skb);
|
|
|
|
int tcp_v4_connect(struct sock *sk, struct sockaddr *uaddr, int addr_len);
|
|
|
|
int tcp_connect(struct sock *sk);
|
2016-04-14 13:05:39 +08:00
|
|
|
enum tcp_synack_type {
|
|
|
|
TCP_SYNACK_NORMAL,
|
|
|
|
TCP_SYNACK_FASTOPEN,
|
|
|
|
TCP_SYNACK_COOKIE,
|
|
|
|
};
|
2015-09-25 22:39:19 +08:00
|
|
|
struct sk_buff *tcp_make_synack(const struct sock *sk, struct dst_entry *dst,
|
2013-09-24 02:33:32 +08:00
|
|
|
struct request_sock *req,
|
2015-10-03 02:43:35 +08:00
|
|
|
struct tcp_fastopen_cookie *foc,
|
2016-04-14 13:05:39 +08:00
|
|
|
enum tcp_synack_type synack_type);
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_disconnect(struct sock *sk, int flags);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2012-04-19 11:40:01 +08:00
|
|
|
void tcp_finish_connect(struct sock *sk, struct sk_buff *skb);
|
2012-05-10 09:49:41 +08:00
|
|
|
int tcp_send_rcvq(struct sock *sk, struct msghdr *msg, size_t size);
|
2012-08-09 22:11:00 +08:00
|
|
|
void inet_sk_rx_dst_set(struct sock *sk, const struct sk_buff *skb);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
|
|
|
/* From syncookies.c */
|
2015-06-05 09:30:43 +08:00
|
|
|
struct sock *tcp_get_cookie_sock(struct sock *sk, struct sk_buff *skb,
|
|
|
|
struct request_sock *req,
|
2017-05-05 21:56:54 +08:00
|
|
|
struct dst_entry *dst, u32 tsoff);
|
2013-09-24 02:33:32 +08:00
|
|
|
int __cookie_v4_check(const struct iphdr *iph, const struct tcphdr *th,
|
|
|
|
u32 cookie);
|
2014-10-16 05:33:22 +08:00
|
|
|
struct sock *cookie_v4_check(struct sock *sk, struct sk_buff *skb);
|
2011-09-19 09:02:55 +08:00
|
|
|
#ifdef CONFIG_SYN_COOKIES
|
2013-09-21 04:32:55 +08:00
|
|
|
|
2014-03-20 12:02:21 +08:00
|
|
|
/* Syncookies use a monotonic timer which increments every 60 seconds.
|
2013-09-21 04:32:55 +08:00
|
|
|
* This counter is used both as a hash input and partially encoded into
|
|
|
|
* the cookie value. A cookie is only validated further if the delta
|
|
|
|
* between the current counter value and the encoded one is less than this,
|
2014-03-20 12:02:21 +08:00
|
|
|
* i.e. a sent cookie is valid only at most for 2*60 seconds (or less if
|
2013-09-21 04:32:55 +08:00
|
|
|
* the counter advances immediately after a cookie is generated).
|
|
|
|
*/
|
2015-05-15 05:26:56 +08:00
|
|
|
#define MAX_SYNCOOKIE_AGE 2
|
|
|
|
#define TCP_SYNCOOKIE_PERIOD (60 * HZ)
|
|
|
|
#define TCP_SYNCOOKIE_VALID (MAX_SYNCOOKIE_AGE * TCP_SYNCOOKIE_PERIOD)
|
|
|
|
|
|
|
|
/* syncookies: remember time of last synqueue overflow
|
|
|
|
* But do not dirty this field too often (once per second is enough)
|
2015-09-29 22:42:49 +08:00
|
|
|
* It is racy as we do not hold a lock, but race is very minor.
|
2015-05-15 05:26:56 +08:00
|
|
|
*/
|
2015-09-29 22:42:49 +08:00
|
|
|
static inline void tcp_synq_overflow(const struct sock *sk)
|
2015-05-15 05:26:56 +08:00
|
|
|
{
|
|
|
|
unsigned long last_overflow = tcp_sk(sk)->rx_opt.ts_recent_stamp;
|
|
|
|
unsigned long now = jiffies;
|
|
|
|
|
|
|
|
if (time_after(now, last_overflow + HZ))
|
|
|
|
tcp_sk(sk)->rx_opt.ts_recent_stamp = now;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* syncookies: no recent synqueue overflow on this listening socket? */
|
|
|
|
static inline bool tcp_synq_no_recent_overflow(const struct sock *sk)
|
|
|
|
{
|
|
|
|
unsigned long last_overflow = tcp_sk(sk)->rx_opt.ts_recent_stamp;
|
|
|
|
|
|
|
|
return time_after(jiffies, last_overflow + TCP_SYNCOOKIE_VALID);
|
|
|
|
}
|
2013-09-21 04:32:55 +08:00
|
|
|
|
|
|
|
static inline u32 tcp_cookie_time(void)
|
|
|
|
{
|
2014-03-20 12:02:21 +08:00
|
|
|
u64 val = get_jiffies_64();
|
|
|
|
|
2015-05-15 05:26:56 +08:00
|
|
|
do_div(val, TCP_SYNCOOKIE_PERIOD);
|
2014-03-20 12:02:21 +08:00
|
|
|
return val;
|
2013-09-21 04:32:55 +08:00
|
|
|
}
|
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
u32 __cookie_v4_init_sequence(const struct iphdr *iph, const struct tcphdr *th,
|
|
|
|
u16 *mssp);
|
2015-09-29 22:42:49 +08:00
|
|
|
__u32 cookie_v4_init_sequence(const struct sk_buff *skb, __u16 *mss);
|
2017-05-17 05:00:14 +08:00
|
|
|
u64 cookie_init_timestamp(struct request_sock *req);
|
syncookies: split cookie_check_timestamp() into two functions
The function cookie_check_timestamp(), both called from IPv4/6 context,
is being used to decode the echoed timestamp from the SYN/ACK into TCP
options used for follow-up communication with the peer.
We can remove ECN handling from that function, split it into a separate
one, and simply rename the original function into cookie_decode_options().
cookie_decode_options() just fills in tcp_option struct based on the
echoed timestamp received from the peer. Anything that fails in this
function will actually discard the request socket.
While this is the natural place for decoding options such as ECN which
commit 172d69e63c7f ("syncookies: add support for ECN") added, we argue
that in particular for ECN handling, it can be checked at a later point
in time as the request sock would actually not need to be dropped from
this, but just ECN support turned off.
Therefore, we split this functionality into cookie_ecn_ok(), which tells
us if the timestamp indicates ECN support AND the tcp_ecn sysctl is enabled.
This prepares for per-route ECN support: just looking at the tcp_ecn sysctl
won't be enough anymore at that point; if the timestamp indicates ECN
and sysctl tcp_ecn == 0, we will also need to check the ECN dst metric.
This would mean adding a route lookup to cookie_check_timestamp(), which
we definitely want to avoid. As we already do a route lookup at a later
point in cookie_{v4,v6}_check(), we can simply make use of that as well
for the new cookie_ecn_ok() function w/o any additional cost.
Joint work with Daniel Borkmann.
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: David S. Miller <davem@davemloft.net>
2014-11-04 00:35:02 +08:00
|
|
|
bool cookie_timestamp_decode(struct tcp_options_received *opt);
|
|
|
|
bool cookie_ecn_ok(const struct tcp_options_received *opt,
|
net: allow setting ecn via routing table
This patch allows to set ECN on a per-route basis in case the sysctl
tcp_ecn is not set to 1. In other words, when ECN is set for specific
routes, it provides a tcp_ecn=1 behaviour for that route while the rest
of the stack acts according to the global settings.
One can use 'ip route change dev $dev $net features ecn' to toggle this.
Having a more fine-grained per-route setting can be beneficial for various
reasons, for example, 1) within data centers, or 2) local ISPs may deploy
ECN support for their own video/streaming services [1], etc.
There was a recent measurement study/paper [2] which scanned the Alexa's
publicly available top million websites list from a vantage point in US,
Europe and Asia:
Half of the Alexa list will now happily use ECN (tcp_ecn=2, most likely
blamed to commit 255cac91c3 ("tcp: extend ECN sysctl to allow server-side
only ECN") ;)); the break in connectivity on-path was found is about
1 in 10,000 cases. Timeouts rather than receiving back RSTs were much
more common in the negotiation phase (and mostly seen in the Alexa
middle band, ranks around 50k-150k): from 12-thousand hosts on which
there _may_ be ECN-linked connection failures, only 79 failed with RST
when _not_ failing with RST when ECN is not requested.
It's unclear though, how much equipment in the wild actually marks CE
when buffers start to fill up.
We thought about a fallback to non-ECN for retransmitted SYNs as another
global option (which could perhaps one day be made default), but as Eric
points out, there's much more work needed to detect broken middleboxes.
Two examples Eric mentioned are buggy firewalls that accept only a single
SYN per flow, and middleboxes that successfully let an ECN flow establish,
but later mark CE for all packets (so cwnd converges to 1).
[1] http://www.ietf.org/proceedings/89/slides/slides-89-tsvarea-1.pdf, p.15
[2] http://ecn.ethz.ch/
Joint work with Daniel Borkmann.
Reference: http://thread.gmane.org/gmane.linux.network/335797
Suggested-by: Hannes Frederic Sowa <hannes@stressinduktion.org>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: David S. Miller <davem@davemloft.net>
2014-11-04 00:35:03 +08:00
|
|
|
const struct net *net, const struct dst_entry *dst);
|
2008-04-10 18:12:40 +08:00
|
|
|
|
2008-02-08 13:49:26 +08:00
|
|
|
/* From net/ipv6/syncookies.c */
|
2013-09-24 02:33:32 +08:00
|
|
|
int __cookie_v6_check(const struct ipv6hdr *iph, const struct tcphdr *th,
|
|
|
|
u32 cookie);
|
|
|
|
struct sock *cookie_v6_check(struct sock *sk, struct sk_buff *skb);
|
syncookies: split cookie_check_timestamp() into two functions
The function cookie_check_timestamp(), both called from IPv4/6 context,
is being used to decode the echoed timestamp from the SYN/ACK into TCP
options used for follow-up communication with the peer.
We can remove ECN handling from that function, split it into a separate
one, and simply rename the original function into cookie_decode_options().
cookie_decode_options() just fills in tcp_option struct based on the
echoed timestamp received from the peer. Anything that fails in this
function will actually discard the request socket.
While this is the natural place for decoding options such as ECN which
commit 172d69e63c7f ("syncookies: add support for ECN") added, we argue
that in particular for ECN handling, it can be checked at a later point
in time as the request sock would actually not need to be dropped from
this, but just ECN support turned off.
Therefore, we split this functionality into cookie_ecn_ok(), which tells
us if the timestamp indicates ECN support AND the tcp_ecn sysctl is enabled.
This prepares for per-route ECN support: just looking at the tcp_ecn sysctl
won't be enough anymore at that point; if the timestamp indicates ECN
and sysctl tcp_ecn == 0, we will also need to check the ECN dst metric.
This would mean adding a route lookup to cookie_check_timestamp(), which
we definitely want to avoid. As we already do a route lookup at a later
point in cookie_{v4,v6}_check(), we can simply make use of that as well
for the new cookie_ecn_ok() function w/o any additional cost.
Joint work with Daniel Borkmann.
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: David S. Miller <davem@davemloft.net>
2014-11-04 00:35:02 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
u32 __cookie_v6_init_sequence(const struct ipv6hdr *iph,
|
|
|
|
const struct tcphdr *th, u16 *mssp);
|
2015-09-29 22:42:49 +08:00
|
|
|
__u32 cookie_v6_init_sequence(const struct sk_buff *skb, __u16 *mss);
|
2011-09-19 09:02:55 +08:00
|
|
|
#endif
|
2005-04-17 06:20:36 +08:00
|
|
|
/* tcp_output.c */
|
|
|
|
|
2016-09-20 11:39:18 +08:00
|
|
|
u32 tcp_tso_autosize(const struct sock *sk, unsigned int mss_now,
|
|
|
|
int min_tso_segs);
|
2013-09-24 02:33:32 +08:00
|
|
|
void __tcp_push_pending_frames(struct sock *sk, unsigned int cur_mss,
|
|
|
|
int nonagle);
|
|
|
|
bool tcp_may_send_now(struct sock *sk);
|
tcp-tso: do not split TSO packets at retransmit time
Linux TCP stack painfully segments all TSO/GSO packets before retransmits.
This was fine back in the days when TSO/GSO were emerging, with their
bugs, but we believe the dark age is over.
Keeping big packets in write queues, but also in stack traversal
has a lot of benefits.
- Less memory overhead, because write queues have less skbs
- Less cpu overhead at ACK processing.
- Better SACK processing, as lot of studies mentioned how
awful linux was at this ;)
- Less cpu overhead to send the rtx packets
(IP stack traversal, netfilter traversal, drivers...)
- Better latencies in presence of losses.
- Smaller spikes in fq like packet schedulers, as retransmits
are not constrained by TCP Small Queues.
1 % packet losses are common today, and at 100Gbit speeds, this
translates to ~80,000 losses per second.
Losses are often correlated, and we see many retransmit events
leading to 1-MSS train of packets, at the time hosts are already
under stress.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-04-22 01:55:23 +08:00
|
|
|
int __tcp_retransmit_skb(struct sock *sk, struct sk_buff *skb, int segs);
|
|
|
|
int tcp_retransmit_skb(struct sock *sk, struct sk_buff *skb, int segs);
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_retransmit_timer(struct sock *sk);
|
|
|
|
void tcp_xmit_retransmit_queue(struct sock *);
|
|
|
|
void tcp_simple_retransmit(struct sock *);
|
2017-01-13 14:11:33 +08:00
|
|
|
void tcp_enter_recovery(struct sock *sk, bool ece_ack);
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_trim_head(struct sock *, struct sk_buff *, u32);
|
2014-06-06 22:32:37 +08:00
|
|
|
int tcp_fragment(struct sock *, struct sk_buff *, u32, unsigned int, gfp_t);
|
2013-09-24 02:33:32 +08:00
|
|
|
|
|
|
|
void tcp_send_probe0(struct sock *);
|
|
|
|
void tcp_send_partial(struct sock *);
|
2015-05-07 05:26:25 +08:00
|
|
|
int tcp_write_wakeup(struct sock *, int mib);
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_send_fin(struct sock *sk);
|
|
|
|
void tcp_send_active_reset(struct sock *sk, gfp_t priority);
|
|
|
|
int tcp_send_synack(struct sock *);
|
|
|
|
void tcp_push_one(struct sock *, unsigned int mss_now);
|
|
|
|
void tcp_send_ack(struct sock *sk);
|
|
|
|
void tcp_send_delayed_ack(struct sock *sk);
|
|
|
|
void tcp_send_loss_probe(struct sock *sk);
|
|
|
|
bool tcp_schedule_loss_probe(struct sock *sk);
|
tcp: Merge tx_flags and tskey in tcp_shifted_skb
After receiving sacks, tcp_shifted_skb() will collapse
skbs if possible. tx_flags and tskey also have to be
merged.
This patch reuses the tcp_skb_collapse_tstamp() to handle
them.
BPF Output Before:
~~~~~
<no-output-due-to-missing-tstamp-event>
BPF Output After:
~~~~~
<...>-2024 [007] d.s. 88.644374: : ee_data:14599
Packetdrill Script:
~~~~~
+0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10`
+0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1`
+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
0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7>
0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7>
0.200 < . 1:1(0) ack 1 win 257
0.200 accept(3, ..., ...) = 4
+0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0
0.200 write(4, ..., 1460) = 1460
+0 setsockopt(4, SOL_SOCKET, 37, [2688], 4) = 0
0.200 write(4, ..., 13140) = 13140
0.200 > P. 1:1461(1460) ack 1
0.200 > . 1461:8761(7300) ack 1
0.200 > P. 8761:14601(5840) ack 1
0.300 < . 1:1(0) ack 1 win 257 <sack 1461:14601,nop,nop>
0.300 > P. 1:1461(1460) ack 1
0.400 < . 1:1(0) ack 14601 win 257
0.400 close(4) = 0
0.400 > F. 14601:14601(0) ack 1
0.500 < F. 1:1(0) ack 14602 win 257
0.500 > . 14602:14602(0) ack 2
Signed-off-by: Martin KaFai Lau <kafai@fb.com>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Soheil Hassas Yeganeh <soheil@google.com>
Cc: Willem de Bruijn <willemb@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Tested-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-04-20 13:39:29 +08:00
|
|
|
void tcp_skb_collapse_tstamp(struct sk_buff *skb,
|
|
|
|
const struct sk_buff *next_skb);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2005-07-06 06:18:51 +08:00
|
|
|
/* tcp_input.c */
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_rearm_rto(struct sock *sk);
|
2015-09-19 02:36:14 +08:00
|
|
|
void tcp_synack_rtt_meas(struct sock *sk, struct request_sock *req);
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_reset(struct sock *sk);
|
2015-10-17 12:57:47 +08:00
|
|
|
void tcp_skb_mark_lost_uncond_verify(struct tcp_sock *tp, struct sk_buff *skb);
|
2016-02-07 03:16:28 +08:00
|
|
|
void tcp_fin(struct sock *sk);
|
2005-07-06 06:18:51 +08:00
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
/* tcp_timer.c */
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_init_xmit_timers(struct sock *);
|
2005-08-10 11:10:42 +08:00
|
|
|
static inline void tcp_clear_xmit_timers(struct sock *sk)
|
|
|
|
{
|
2017-05-16 19:24:36 +08:00
|
|
|
hrtimer_cancel(&tcp_sk(sk)->pacing_timer);
|
2005-08-10 11:10:42 +08:00
|
|
|
inet_csk_clear_xmit_timers(sk);
|
|
|
|
}
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
unsigned int tcp_sync_mss(struct sock *sk, u32 pmtu);
|
|
|
|
unsigned int tcp_current_mss(struct sock *sk);
|
2009-03-14 22:23:05 +08:00
|
|
|
|
|
|
|
/* Bound MSS / TSO packet size with the half of the window */
|
|
|
|
static inline int tcp_bound_to_half_wnd(struct tcp_sock *tp, int pktsize)
|
|
|
|
{
|
2010-09-16 01:27:52 +08:00
|
|
|
int cutoff;
|
|
|
|
|
|
|
|
/* When peer uses tiny windows, there is no use in packetizing
|
|
|
|
* to sub-MSS pieces for the sake of SWS or making sure there
|
|
|
|
* are enough packets in the pipe for fast recovery.
|
|
|
|
*
|
|
|
|
* On the other hand, for extremely large MSS devices, handling
|
|
|
|
* smaller than MSS windows in this way does make sense.
|
|
|
|
*/
|
2016-06-29 07:06:48 +08:00
|
|
|
if (tp->max_window > TCP_MSS_DEFAULT)
|
2010-09-16 01:27:52 +08:00
|
|
|
cutoff = (tp->max_window >> 1);
|
|
|
|
else
|
|
|
|
cutoff = tp->max_window;
|
|
|
|
|
|
|
|
if (cutoff && pktsize > cutoff)
|
|
|
|
return max_t(int, cutoff, 68U - tp->tcp_header_len);
|
2009-03-14 22:23:05 +08:00
|
|
|
else
|
|
|
|
return pktsize;
|
|
|
|
}
|
2005-04-17 06:20:36 +08:00
|
|
|
|
[INET_DIAG]: Move the tcp_diag interface to the proper place
With this the previous setup is back, i.e. tcp_diag can be built as a module,
as dccp_diag and both share the infrastructure available in inet_diag.
If one selects CONFIG_INET_DIAG as module CONFIG_INET_TCP_DIAG will also be
built as a module, as will CONFIG_INET_DCCP_DIAG, if CONFIG_IP_DCCP was
selected static or as a module, if CONFIG_INET_DIAG is y, being statically
linked CONFIG_INET_TCP_DIAG will follow suit and CONFIG_INET_DCCP_DIAG will be
built in the same manner as CONFIG_IP_DCCP.
Now to aim at UDP, converting it to use inet_hashinfo, so that we can use
iproute2 for UDP sockets as well.
Ah, just to show an example of this new infrastructure working for DCCP :-)
[root@qemu ~]# ./ss -dane
State Recv-Q Send-Q Local Address:Port Peer Address:Port
LISTEN 0 0 *:5001 *:* ino:942 sk:cfd503a0
ESTAB 0 0 127.0.0.1:5001 127.0.0.1:32770 ino:943 sk:cfd50a60
ESTAB 0 0 127.0.0.1:32770 127.0.0.1:5001 ino:947 sk:cfd50700
TIME-WAIT 0 0 127.0.0.1:32769 127.0.0.1:5001 timer:(timewait,3.430ms,0) ino:0 sk:cf209620
Signed-off-by: Arnaldo Carvalho de Melo <acme@mandriva.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2005-08-12 23:59:17 +08:00
|
|
|
/* tcp.c */
|
2015-04-29 06:28:17 +08:00
|
|
|
void tcp_get_info(struct sock *, struct tcp_info *);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
|
|
|
/* Read 'sendfile()'-style from a TCP socket */
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_read_sock(struct sock *sk, read_descriptor_t *desc,
|
|
|
|
sk_read_actor_t recv_actor);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_initialize_rcv_mss(struct sock *sk);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_mtu_to_mss(struct sock *sk, int pmtu);
|
|
|
|
int tcp_mss_to_mtu(struct sock *sk, int mss);
|
|
|
|
void tcp_mtup_init(struct sock *sk);
|
|
|
|
void tcp_init_buffer_space(struct sock *sk);
|
2006-03-21 09:53:41 +08:00
|
|
|
|
Revert Backoff [v3]: Revert RTO on ICMP destination unreachable
Here, an ICMP host/network unreachable message, whose payload fits to
TCP's SND.UNA, is taken as an indication that the RTO retransmission has
not been lost due to congestion, but because of a route failure
somewhere along the path.
With true congestion, a router won't trigger such a message and the
patched TCP will operate as standard TCP.
This patch reverts one RTO backoff, if an ICMP host/network unreachable
message, whose payload fits to TCP's SND.UNA, arrives.
Based on the new RTO, the retransmission timer is reset to reflect the
remaining time, or - if the revert clocked out the timer - a retransmission
is sent out immediately.
Backoffs are only reverted, if TCP is in RTO loss recovery, i.e. if
there have been retransmissions and reversible backoffs, already.
Changes from v2:
1) Renaming of skb in tcp_v4_err() moved to another patch.
2) Reintroduced tcp_bound_rto() and __tcp_set_rto().
3) Fixed code comments.
Signed-off-by: Damian Lukowski <damian@tvk.rwth-aachen.de>
Acked-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
2009-08-26 08:16:31 +08:00
|
|
|
static inline void tcp_bound_rto(const struct sock *sk)
|
|
|
|
{
|
|
|
|
if (inet_csk(sk)->icsk_rto > TCP_RTO_MAX)
|
|
|
|
inet_csk(sk)->icsk_rto = TCP_RTO_MAX;
|
|
|
|
}
|
|
|
|
|
|
|
|
static inline u32 __tcp_set_rto(const struct tcp_sock *tp)
|
|
|
|
{
|
2014-02-27 06:02:48 +08:00
|
|
|
return usecs_to_jiffies((tp->srtt_us >> 3) + tp->rttvar_us);
|
Revert Backoff [v3]: Revert RTO on ICMP destination unreachable
Here, an ICMP host/network unreachable message, whose payload fits to
TCP's SND.UNA, is taken as an indication that the RTO retransmission has
not been lost due to congestion, but because of a route failure
somewhere along the path.
With true congestion, a router won't trigger such a message and the
patched TCP will operate as standard TCP.
This patch reverts one RTO backoff, if an ICMP host/network unreachable
message, whose payload fits to TCP's SND.UNA, arrives.
Based on the new RTO, the retransmission timer is reset to reflect the
remaining time, or - if the revert clocked out the timer - a retransmission
is sent out immediately.
Backoffs are only reverted, if TCP is in RTO loss recovery, i.e. if
there have been retransmissions and reversible backoffs, already.
Changes from v2:
1) Renaming of skb in tcp_v4_err() moved to another patch.
2) Reintroduced tcp_bound_rto() and __tcp_set_rto().
3) Fixed code comments.
Signed-off-by: Damian Lukowski <damian@tvk.rwth-aachen.de>
Acked-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
2009-08-26 08:16:31 +08:00
|
|
|
}
|
|
|
|
|
2006-01-04 08:03:49 +08:00
|
|
|
static inline void __tcp_fast_path_on(struct tcp_sock *tp, u32 snd_wnd)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
|
|
|
tp->pred_flags = htonl((tp->tcp_header_len << 26) |
|
|
|
|
ntohl(TCP_FLAG_ACK) |
|
|
|
|
snd_wnd);
|
|
|
|
}
|
|
|
|
|
2006-01-04 08:03:49 +08:00
|
|
|
static inline void tcp_fast_path_on(struct tcp_sock *tp)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
|
|
|
__tcp_fast_path_on(tp, tp->snd_wnd >> tp->rx_opt.snd_wscale);
|
|
|
|
}
|
|
|
|
|
[TCP]: Sed magic converts func(sk, tp, ...) -> func(sk, ...)
This is (mostly) automated change using magic:
sed -e '/struct sock \*sk/ N' -e '/struct sock \*sk/ N'
-e '/struct sock \*sk/ N' -e '/struct sock \*sk/ N'
-e 's|struct sock \*sk,[\n\t ]*struct tcp_sock \*tp\([^{]*\n{\n\)|
struct sock \*sk\1\tstruct tcp_sock *tp = tcp_sk(sk);\n|g'
-e 's|struct sock \*sk, struct tcp_sock \*tp|
struct sock \*sk|g' -e 's|sk, tp\([^-]\)|sk\1|g'
Fixed four unused variable (tp) warnings that were introduced.
In addition, manually added newlines after local variables and
tweaked function arguments positioning.
$ gcc --version
gcc (GCC) 4.1.1 20060525 (Red Hat 4.1.1-1)
...
$ codiff -fV built-in.o.old built-in.o.new
net/ipv4/route.c:
rt_cache_flush | +14
1 function changed, 14 bytes added
net/ipv4/tcp.c:
tcp_setsockopt | -5
tcp_sendpage | -25
tcp_sendmsg | -16
3 functions changed, 46 bytes removed
net/ipv4/tcp_input.c:
tcp_try_undo_recovery | +3
tcp_try_undo_dsack | +2
tcp_mark_head_lost | -12
tcp_ack | -15
tcp_event_data_recv | -32
tcp_rcv_state_process | -10
tcp_rcv_established | +1
7 functions changed, 6 bytes added, 69 bytes removed, diff: -63
net/ipv4/tcp_output.c:
update_send_head | -9
tcp_transmit_skb | +19
tcp_cwnd_validate | +1
tcp_write_wakeup | -17
__tcp_push_pending_frames | -25
tcp_push_one | -8
tcp_send_fin | -4
7 functions changed, 20 bytes added, 63 bytes removed, diff: -43
built-in.o.new:
18 functions changed, 40 bytes added, 178 bytes removed, diff: -138
Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
2007-04-21 13:18:02 +08:00
|
|
|
static inline void tcp_fast_path_check(struct sock *sk)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
[TCP]: Sed magic converts func(sk, tp, ...) -> func(sk, ...)
This is (mostly) automated change using magic:
sed -e '/struct sock \*sk/ N' -e '/struct sock \*sk/ N'
-e '/struct sock \*sk/ N' -e '/struct sock \*sk/ N'
-e 's|struct sock \*sk,[\n\t ]*struct tcp_sock \*tp\([^{]*\n{\n\)|
struct sock \*sk\1\tstruct tcp_sock *tp = tcp_sk(sk);\n|g'
-e 's|struct sock \*sk, struct tcp_sock \*tp|
struct sock \*sk|g' -e 's|sk, tp\([^-]\)|sk\1|g'
Fixed four unused variable (tp) warnings that were introduced.
In addition, manually added newlines after local variables and
tweaked function arguments positioning.
$ gcc --version
gcc (GCC) 4.1.1 20060525 (Red Hat 4.1.1-1)
...
$ codiff -fV built-in.o.old built-in.o.new
net/ipv4/route.c:
rt_cache_flush | +14
1 function changed, 14 bytes added
net/ipv4/tcp.c:
tcp_setsockopt | -5
tcp_sendpage | -25
tcp_sendmsg | -16
3 functions changed, 46 bytes removed
net/ipv4/tcp_input.c:
tcp_try_undo_recovery | +3
tcp_try_undo_dsack | +2
tcp_mark_head_lost | -12
tcp_ack | -15
tcp_event_data_recv | -32
tcp_rcv_state_process | -10
tcp_rcv_established | +1
7 functions changed, 6 bytes added, 69 bytes removed, diff: -63
net/ipv4/tcp_output.c:
update_send_head | -9
tcp_transmit_skb | +19
tcp_cwnd_validate | +1
tcp_write_wakeup | -17
__tcp_push_pending_frames | -25
tcp_push_one | -8
tcp_send_fin | -4
7 functions changed, 20 bytes added, 63 bytes removed, diff: -43
built-in.o.new:
18 functions changed, 40 bytes added, 178 bytes removed, diff: -138
Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
2007-04-21 13:18:02 +08:00
|
|
|
struct tcp_sock *tp = tcp_sk(sk);
|
|
|
|
|
tcp: use an RB tree for ooo receive queue
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: Yaogong Wang <wygivan@google.com>
Signed-off-by: Eric 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: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-08 05:49:28 +08:00
|
|
|
if (RB_EMPTY_ROOT(&tp->out_of_order_queue) &&
|
2005-04-17 06:20:36 +08:00
|
|
|
tp->rcv_wnd &&
|
|
|
|
atomic_read(&sk->sk_rmem_alloc) < sk->sk_rcvbuf &&
|
|
|
|
!tp->urg_data)
|
|
|
|
tcp_fast_path_on(tp);
|
|
|
|
}
|
|
|
|
|
2009-05-05 02:11:01 +08:00
|
|
|
/* Compute the actual rto_min value */
|
|
|
|
static inline u32 tcp_rto_min(struct sock *sk)
|
|
|
|
{
|
2011-10-21 17:22:42 +08:00
|
|
|
const struct dst_entry *dst = __sk_dst_get(sk);
|
2009-05-05 02:11:01 +08:00
|
|
|
u32 rto_min = TCP_RTO_MIN;
|
|
|
|
|
|
|
|
if (dst && dst_metric_locked(dst, RTAX_RTO_MIN))
|
|
|
|
rto_min = dst_metric_rtt(dst, RTAX_RTO_MIN);
|
|
|
|
return rto_min;
|
|
|
|
}
|
|
|
|
|
2014-02-27 06:02:48 +08:00
|
|
|
static inline u32 tcp_rto_min_us(struct sock *sk)
|
|
|
|
{
|
|
|
|
return jiffies_to_usecs(tcp_rto_min(sk));
|
|
|
|
}
|
|
|
|
|
net: tcp: add per route congestion control
This work adds the possibility to define a per route/destination
congestion control algorithm. Generally, this opens up the possibility
for a machine with different links to enforce specific congestion
control algorithms with optimal strategies for each of them based
on their network characteristics, even transparently for a single
application listening on all links.
For our specific use case, this additionally facilitates deployment
of DCTCP, for example, applications can easily serve internal
traffic/dsts in DCTCP and external one with CUBIC. Other scenarios
would also allow for utilizing e.g. long living, low priority
background flows for certain destinations/routes while still being
able for normal traffic to utilize the default congestion control
algorithm. We also thought about a per netns setting (where different
defaults are possible), but given its actually a link specific
property, we argue that a per route/destination setting is the most
natural and flexible.
The administrator can utilize this through ip-route(8) by appending
"congctl [lock] <name>", where <name> denotes the name of a
congestion control algorithm and the optional lock parameter allows
to enforce the given algorithm so that applications in user space
would not be allowed to overwrite that algorithm for that destination.
The dst metric lookups are being done when a dst entry is already
available in order to avoid a costly lookup and still before the
algorithms are being initialized, thus overhead is very low when the
feature is not being used. While the client side would need to drop
the current reference on the module, on server side this can actually
even be avoided as we just got a flat-copied socket clone.
Joint work with Florian Westphal.
Suggested-by: Hannes Frederic Sowa <hannes@stressinduktion.org>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:48 +08:00
|
|
|
static inline bool tcp_ca_dst_locked(const struct dst_entry *dst)
|
|
|
|
{
|
|
|
|
return dst_metric_locked(dst, RTAX_CC_ALGO);
|
|
|
|
}
|
|
|
|
|
2015-10-17 12:57:42 +08:00
|
|
|
/* Minimum RTT in usec. ~0 means not available. */
|
|
|
|
static inline u32 tcp_min_rtt(const struct tcp_sock *tp)
|
|
|
|
{
|
2016-09-20 11:39:10 +08:00
|
|
|
return minmax_get(&tp->rtt_min);
|
2015-10-17 12:57:42 +08:00
|
|
|
}
|
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
/* Compute the actual receive window we are currently advertising.
|
|
|
|
* Rcv_nxt can be after the window if our peer push more data
|
|
|
|
* than the offered window.
|
|
|
|
*/
|
2006-01-04 08:03:49 +08:00
|
|
|
static inline u32 tcp_receive_window(const struct tcp_sock *tp)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
|
|
|
s32 win = tp->rcv_wup + tp->rcv_wnd - tp->rcv_nxt;
|
|
|
|
|
|
|
|
if (win < 0)
|
|
|
|
win = 0;
|
|
|
|
return (u32) win;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Choose a new window, without checks for shrinking, and without
|
|
|
|
* scaling applied to the result. The caller does these things
|
|
|
|
* if necessary. This is a "raw" window selection.
|
|
|
|
*/
|
2013-09-24 02:33:32 +08:00
|
|
|
u32 __tcp_select_window(struct sock *sk);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2012-04-19 11:40:39 +08:00
|
|
|
void tcp_send_window_probe(struct sock *sk);
|
|
|
|
|
2017-05-17 05:00:01 +08:00
|
|
|
/* TCP uses 32bit jiffies to save some space.
|
|
|
|
* Note that this is different from tcp_time_stamp, which
|
|
|
|
* historically has been the same until linux-4.13.
|
|
|
|
*/
|
|
|
|
#define tcp_jiffies32 ((u32)jiffies)
|
|
|
|
|
2017-05-17 05:00:14 +08:00
|
|
|
/*
|
|
|
|
* Deliver a 32bit value for TCP timestamp option (RFC 7323)
|
|
|
|
* It is no longer tied to jiffies, but to 1 ms clock.
|
|
|
|
* Note: double check if you want to use tcp_jiffies32 instead of this.
|
|
|
|
*/
|
|
|
|
#define TCP_TS_HZ 1000
|
|
|
|
|
|
|
|
static inline u64 tcp_clock_ns(void)
|
|
|
|
{
|
|
|
|
return local_clock();
|
|
|
|
}
|
|
|
|
|
|
|
|
static inline u64 tcp_clock_us(void)
|
|
|
|
{
|
|
|
|
return div_u64(tcp_clock_ns(), NSEC_PER_USEC);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* This should only be used in contexts where tp->tcp_mstamp is up to date */
|
|
|
|
static inline u32 tcp_time_stamp(const struct tcp_sock *tp)
|
|
|
|
{
|
|
|
|
return div_u64(tp->tcp_mstamp, USEC_PER_SEC / TCP_TS_HZ);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Could use tcp_clock_us() / 1000, but this version uses a single divide */
|
|
|
|
static inline u32 tcp_time_stamp_raw(void)
|
|
|
|
{
|
|
|
|
return div_u64(tcp_clock_ns(), NSEC_PER_SEC / TCP_TS_HZ);
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
/* Refresh 1us clock of a TCP socket,
|
|
|
|
* ensuring monotically increasing values.
|
2005-04-17 06:20:36 +08:00
|
|
|
*/
|
2017-05-17 05:00:14 +08:00
|
|
|
static inline void tcp_mstamp_refresh(struct tcp_sock *tp)
|
|
|
|
{
|
|
|
|
u64 val = tcp_clock_us();
|
|
|
|
|
|
|
|
if (val > tp->tcp_mstamp)
|
|
|
|
tp->tcp_mstamp = val;
|
|
|
|
}
|
|
|
|
|
|
|
|
static inline u32 tcp_stamp_us_delta(u64 t1, u64 t0)
|
|
|
|
{
|
|
|
|
return max_t(s64, t1 - t0, 0);
|
|
|
|
}
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2014-09-06 06:33:33 +08:00
|
|
|
static inline u32 tcp_skb_timestamp(const struct sk_buff *skb)
|
|
|
|
{
|
2017-05-17 05:00:14 +08:00
|
|
|
return div_u64(skb->skb_mstamp, USEC_PER_SEC / TCP_TS_HZ);
|
2014-09-06 06:33:33 +08:00
|
|
|
}
|
|
|
|
|
|
|
|
|
2010-06-12 22:01:43 +08:00
|
|
|
#define tcp_flag_byte(th) (((u_int8_t *)th)[13])
|
|
|
|
|
|
|
|
#define TCPHDR_FIN 0x01
|
|
|
|
#define TCPHDR_SYN 0x02
|
|
|
|
#define TCPHDR_RST 0x04
|
|
|
|
#define TCPHDR_PSH 0x08
|
|
|
|
#define TCPHDR_ACK 0x10
|
|
|
|
#define TCPHDR_URG 0x20
|
|
|
|
#define TCPHDR_ECE 0x40
|
|
|
|
#define TCPHDR_CWR 0x80
|
|
|
|
|
tcp: add rfc3168, section 6.1.1.1. fallback
This work as a follow-up of commit f7b3bec6f516 ("net: allow setting ecn
via routing table") and adds RFC3168 section 6.1.1.1. fallback for outgoing
ECN connections. In other words, this work adds a retry with a non-ECN
setup SYN packet, as suggested from the RFC on the first timeout:
[...] A host that receives no reply to an ECN-setup SYN within the
normal SYN retransmission timeout interval MAY resend the SYN and
any subsequent SYN retransmissions with CWR and ECE cleared. [...]
Schematic client-side view when assuming the server is in tcp_ecn=2 mode,
that is, Linux default since 2009 via commit 255cac91c3c9 ("tcp: extend
ECN sysctl to allow server-side only ECN"):
1) Normal ECN-capable path:
SYN ECE CWR ----->
<----- SYN ACK ECE
ACK ----->
2) Path with broken middlebox, when client has fallback:
SYN ECE CWR ----X crappy middlebox drops packet
(timeout, rtx)
SYN ----->
<----- SYN ACK
ACK ----->
In case we would not have the fallback implemented, the middlebox drop
point would basically end up as:
SYN ECE CWR ----X crappy middlebox drops packet
(timeout, rtx)
SYN ECE CWR ----X crappy middlebox drops packet
(timeout, rtx)
SYN ECE CWR ----X crappy middlebox drops packet
(timeout, rtx)
In any case, it's rather a smaller percentage of sites where there would
occur such additional setup latency: it was found in end of 2014 that ~56%
of IPv4 and 65% of IPv6 servers of Alexa 1 million list would negotiate
ECN (aka tcp_ecn=2 default), 0.42% of these webservers will fail to connect
when trying to negotiate with ECN (tcp_ecn=1) due to timeouts, which the
fallback would mitigate with a slight latency trade-off. Recent related
paper on this topic:
Brian Trammell, Mirja Kühlewind, Damiano Boppart, Iain Learmonth,
Gorry Fairhurst, and Richard Scheffenegger:
"Enabling Internet-Wide Deployment of Explicit Congestion Notification."
Proc. PAM 2015, New York.
http://ecn.ethz.ch/ecn-pam15.pdf
Thus, when net.ipv4.tcp_ecn=1 is being set, the patch will perform RFC3168,
section 6.1.1.1. fallback on timeout. For users explicitly not wanting this
which can be in DC use case, we add a net.ipv4.tcp_ecn_fallback knob that
allows for disabling the fallback.
tp->ecn_flags are not being cleared in tcp_ecn_clear_syn() on output, but
rather we let tcp_ecn_rcv_synack() take that over on input path in case a
SYN ACK ECE was delayed. Thus a spurious SYN retransmission will not prevent
ECN being negotiated eventually in that case.
Reference: https://www.ietf.org/proceedings/92/slides/slides-92-iccrg-1.pdf
Reference: https://www.ietf.org/proceedings/89/slides/slides-89-tsvarea-1.pdf
Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Mirja Kühlewind <mirja.kuehlewind@tik.ee.ethz.ch>
Signed-off-by: Brian Trammell <trammell@tik.ee.ethz.ch>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Dave That <dave.taht@gmail.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-05-20 03:04:22 +08:00
|
|
|
#define TCPHDR_SYN_ECN (TCPHDR_SYN | TCPHDR_ECE | TCPHDR_CWR)
|
|
|
|
|
2005-11-11 09:13:47 +08:00
|
|
|
/* This is what the send packet queuing engine uses to pass
|
2010-07-16 12:41:00 +08:00
|
|
|
* TCP per-packet control information to the transmission code.
|
|
|
|
* We also store the host-order sequence numbers in here too.
|
|
|
|
* This is 44 bytes if IPV6 is enabled.
|
|
|
|
* If this grows please adjust skbuff.h:skbuff->cb[xxx] size appropriately.
|
2005-04-17 06:20:36 +08:00
|
|
|
*/
|
|
|
|
struct tcp_skb_cb {
|
|
|
|
__u32 seq; /* Starting sequence number */
|
|
|
|
__u32 end_seq; /* SEQ + FIN + SYN + datalen */
|
2014-09-24 19:11:22 +08:00
|
|
|
union {
|
|
|
|
/* Note : tcp_tw_isn is used in input path only
|
|
|
|
* (isn chosen by tcp_timewait_state_process())
|
|
|
|
*
|
2015-06-12 00:15:18 +08:00
|
|
|
* tcp_gso_segs/size are used in write queue only,
|
|
|
|
* cf tcp_skb_pcount()/tcp_skb_mss()
|
2014-09-24 19:11:22 +08:00
|
|
|
*/
|
|
|
|
__u32 tcp_tw_isn;
|
2015-06-12 00:15:18 +08:00
|
|
|
struct {
|
|
|
|
u16 tcp_gso_segs;
|
|
|
|
u16 tcp_gso_size;
|
|
|
|
};
|
2014-09-24 19:11:22 +08:00
|
|
|
};
|
2011-09-28 01:25:05 +08:00
|
|
|
__u8 tcp_flags; /* TCP header flags. (tcp[13]) */
|
2012-04-16 15:08:06 +08:00
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
__u8 sacked; /* State flags for SACK/FACK. */
|
|
|
|
#define TCPCB_SACKED_ACKED 0x01 /* SKB ACK'd by a SACK block */
|
|
|
|
#define TCPCB_SACKED_RETRANS 0x02 /* SKB retransmitted */
|
|
|
|
#define TCPCB_LOST 0x04 /* SKB is lost */
|
|
|
|
#define TCPCB_TAGBITS 0x07 /* All tag bits */
|
2014-08-13 20:03:10 +08:00
|
|
|
#define TCPCB_REPAIRED 0x10 /* SKB repaired (no skb_mstamp) */
|
2005-04-17 06:20:36 +08:00
|
|
|
#define TCPCB_EVER_RETRANS 0x80 /* Ever retransmitted frame */
|
2014-08-13 20:03:10 +08:00
|
|
|
#define TCPCB_RETRANS (TCPCB_SACKED_RETRANS|TCPCB_EVER_RETRANS| \
|
|
|
|
TCPCB_REPAIRED)
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2012-04-16 15:08:06 +08:00
|
|
|
__u8 ip_dsfield; /* IPv4 tos or IPv6 dsfield */
|
2016-04-03 11:08:08 +08:00
|
|
|
__u8 txstamp_ack:1, /* Record TX timestamp for ack? */
|
tcp: Make use of MSG_EOR in tcp_sendmsg
This patch adds an eor bit to the TCP_SKB_CB. When MSG_EOR
is passed to tcp_sendmsg, the eor bit will be set at the skb
containing the last byte of the userland's msg. The eor bit
will prevent data from appending to that skb in the future.
The change in do_tcp_sendpages is to honor the eor set
during the previous tcp_sendmsg(MSG_EOR) call.
This patch handles the tcp_sendmsg case. The followup patches
will handle other skb coalescing and fragment cases.
One potential use case is to use MSG_EOR with
SOF_TIMESTAMPING_TX_ACK to get a more accurate
TCP ack timestamping on application protocol with
multiple outgoing response messages (e.g. HTTP2).
Packetdrill script for testing:
~~~~~~
+0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10`
+0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1`
+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
0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7>
0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7>
0.200 < . 1:1(0) ack 1 win 257
0.200 accept(3, ..., ...) = 4
+0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0
0.200 write(4, ..., 14600) = 14600
0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730
0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730
0.200 > . 1:7301(7300) ack 1
0.200 > P. 7301:14601(7300) ack 1
0.300 < . 1:1(0) ack 14601 win 257
0.300 > P. 14601:15331(730) ack 1
0.300 > P. 15331:16061(730) ack 1
0.400 < . 1:1(0) ack 16061 win 257
0.400 close(4) = 0
0.400 > F. 16061:16061(0) ack 1
0.400 < F. 1:1(0) ack 16062 win 257
0.400 > . 16062:16062(0) ack 2
Signed-off-by: Martin KaFai Lau <kafai@fb.com>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Soheil Hassas Yeganeh <soheil@google.com>
Cc: Willem de Bruijn <willemb@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Suggested-by: Eric Dumazet <edumazet@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-04-26 05:44:48 +08:00
|
|
|
eor:1, /* Is skb MSG_EOR marked? */
|
|
|
|
unused:6;
|
2005-04-17 06:20:36 +08:00
|
|
|
__u32 ack_seq; /* Sequence number ACK'd */
|
2014-09-28 00:50:57 +08:00
|
|
|
union {
|
2016-05-07 11:35:35 +08:00
|
|
|
struct {
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
|
|
|
/* There is space for up to 24 bytes */
|
2016-09-20 11:39:15 +08:00
|
|
|
__u32 in_flight:30,/* Bytes in flight at transmit */
|
|
|
|
is_app_limited:1, /* cwnd not fully used? */
|
|
|
|
unused:1;
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
|
|
|
/* pkts S/ACKed so far upon tx of skb, incl retrans: */
|
|
|
|
__u32 delivered;
|
|
|
|
/* start of send pipeline phase */
|
2017-05-17 05:00:14 +08:00
|
|
|
u64 first_tx_mstamp;
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
|
|
|
/* when we reached the "delivered" count */
|
2017-05-17 05:00:14 +08:00
|
|
|
u64 delivered_mstamp;
|
2016-05-07 11:35:35 +08:00
|
|
|
} tx; /* only used for outgoing skbs */
|
|
|
|
union {
|
|
|
|
struct inet_skb_parm h4;
|
2014-09-28 00:50:57 +08:00
|
|
|
#if IS_ENABLED(CONFIG_IPV6)
|
2016-05-07 11:35:35 +08:00
|
|
|
struct inet6_skb_parm h6;
|
2014-09-28 00:50:57 +08:00
|
|
|
#endif
|
2016-05-07 11:35:35 +08:00
|
|
|
} header; /* For incoming skbs */
|
|
|
|
};
|
2005-04-17 06:20:36 +08:00
|
|
|
};
|
|
|
|
|
|
|
|
#define TCP_SKB_CB(__skb) ((struct tcp_skb_cb *)&((__skb)->cb[0]))
|
|
|
|
|
2014-10-18 00:17:20 +08:00
|
|
|
|
2014-10-18 23:34:37 +08:00
|
|
|
#if IS_ENABLED(CONFIG_IPV6)
|
2014-10-18 00:17:20 +08:00
|
|
|
/* This is the variant of inet6_iif() that must be used by TCP,
|
|
|
|
* as TCP moves IP6CB into a different location in skb->cb[]
|
|
|
|
*/
|
|
|
|
static inline int tcp_v6_iif(const struct sk_buff *skb)
|
|
|
|
{
|
2016-10-17 11:02:52 +08:00
|
|
|
bool l3_slave = ipv6_l3mdev_skb(TCP_SKB_CB(skb)->header.h6.flags);
|
2016-05-11 02:19:50 +08:00
|
|
|
|
|
|
|
return l3_slave ? skb->skb_iif : TCP_SKB_CB(skb)->header.h6.iif;
|
2014-10-18 00:17:20 +08:00
|
|
|
}
|
2014-10-18 23:34:37 +08:00
|
|
|
#endif
|
2014-10-18 00:17:20 +08:00
|
|
|
|
2016-10-17 11:02:52 +08:00
|
|
|
/* TCP_SKB_CB reference means this can not be used from early demux */
|
|
|
|
static inline bool inet_exact_dif_match(struct net *net, struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
#if IS_ENABLED(CONFIG_NET_L3_MASTER_DEV)
|
|
|
|
if (!net->ipv4.sysctl_tcp_l3mdev_accept &&
|
2016-11-03 03:08:25 +08:00
|
|
|
skb && ipv4_l3mdev_skb(TCP_SKB_CB(skb)->header.h4.flags))
|
2016-10-17 11:02:52 +08:00
|
|
|
return true;
|
|
|
|
#endif
|
|
|
|
return false;
|
|
|
|
}
|
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
/* Due to TSO, an SKB can be composed of multiple actual
|
|
|
|
* packets. To keep these tracked properly, we use this.
|
2012-05-04 13:14:02 +08:00
|
|
|
*/
|
2005-04-17 06:20:36 +08:00
|
|
|
static inline int tcp_skb_pcount(const struct sk_buff *skb)
|
2012-05-04 13:14:02 +08:00
|
|
|
{
|
2014-09-24 19:11:22 +08:00
|
|
|
return TCP_SKB_CB(skb)->tcp_gso_segs;
|
|
|
|
}
|
2012-05-04 13:14:02 +08:00
|
|
|
|
2014-09-24 19:11:22 +08:00
|
|
|
static inline void tcp_skb_pcount_set(struct sk_buff *skb, int segs)
|
|
|
|
{
|
|
|
|
TCP_SKB_CB(skb)->tcp_gso_segs = segs;
|
2012-05-04 13:14:02 +08:00
|
|
|
}
|
|
|
|
|
2014-09-24 19:11:22 +08:00
|
|
|
static inline void tcp_skb_pcount_add(struct sk_buff *skb, int segs)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
2014-09-24 19:11:22 +08:00
|
|
|
TCP_SKB_CB(skb)->tcp_gso_segs += segs;
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
2015-06-12 00:15:18 +08:00
|
|
|
/* This is valid iff skb is in write queue and tcp_skb_pcount() > 1. */
|
2005-04-17 06:20:36 +08:00
|
|
|
static inline int tcp_skb_mss(const struct sk_buff *skb)
|
|
|
|
{
|
2015-06-12 00:15:18 +08:00
|
|
|
return TCP_SKB_CB(skb)->tcp_gso_size;
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
tcp: Make use of MSG_EOR in tcp_sendmsg
This patch adds an eor bit to the TCP_SKB_CB. When MSG_EOR
is passed to tcp_sendmsg, the eor bit will be set at the skb
containing the last byte of the userland's msg. The eor bit
will prevent data from appending to that skb in the future.
The change in do_tcp_sendpages is to honor the eor set
during the previous tcp_sendmsg(MSG_EOR) call.
This patch handles the tcp_sendmsg case. The followup patches
will handle other skb coalescing and fragment cases.
One potential use case is to use MSG_EOR with
SOF_TIMESTAMPING_TX_ACK to get a more accurate
TCP ack timestamping on application protocol with
multiple outgoing response messages (e.g. HTTP2).
Packetdrill script for testing:
~~~~~~
+0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10`
+0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1`
+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
0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7>
0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7>
0.200 < . 1:1(0) ack 1 win 257
0.200 accept(3, ..., ...) = 4
+0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0
0.200 write(4, ..., 14600) = 14600
0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730
0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730
0.200 > . 1:7301(7300) ack 1
0.200 > P. 7301:14601(7300) ack 1
0.300 < . 1:1(0) ack 14601 win 257
0.300 > P. 14601:15331(730) ack 1
0.300 > P. 15331:16061(730) ack 1
0.400 < . 1:1(0) ack 16061 win 257
0.400 close(4) = 0
0.400 > F. 16061:16061(0) ack 1
0.400 < F. 1:1(0) ack 16062 win 257
0.400 > . 16062:16062(0) ack 2
Signed-off-by: Martin KaFai Lau <kafai@fb.com>
Cc: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Soheil Hassas Yeganeh <soheil@google.com>
Cc: Willem de Bruijn <willemb@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Suggested-by: Eric Dumazet <edumazet@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-04-26 05:44:48 +08:00
|
|
|
static inline bool tcp_skb_can_collapse_to(const struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
return likely(!TCP_SKB_CB(skb)->eor);
|
|
|
|
}
|
|
|
|
|
2005-06-24 03:19:55 +08:00
|
|
|
/* Events passed to congestion control interface */
|
|
|
|
enum tcp_ca_event {
|
|
|
|
CA_EVENT_TX_START, /* first transmit when no packets in flight */
|
|
|
|
CA_EVENT_CWND_RESTART, /* congestion window restart */
|
|
|
|
CA_EVENT_COMPLETE_CWR, /* end of congestion recovery */
|
|
|
|
CA_EVENT_LOSS, /* loss timeout */
|
2014-09-27 04:37:35 +08:00
|
|
|
CA_EVENT_ECN_NO_CE, /* ECT set, but not CE marked */
|
|
|
|
CA_EVENT_ECN_IS_CE, /* received CE marked IP packet */
|
|
|
|
CA_EVENT_DELAYED_ACK, /* Delayed ack is sent */
|
|
|
|
CA_EVENT_NON_DELAYED_ACK,
|
2014-09-27 04:37:34 +08:00
|
|
|
};
|
|
|
|
|
2014-09-27 04:37:35 +08:00
|
|
|
/* Information about inbound ACK, passed to cong_ops->in_ack_event() */
|
2014-09-27 04:37:34 +08:00
|
|
|
enum tcp_ca_ack_event_flags {
|
2014-09-27 04:37:35 +08:00
|
|
|
CA_ACK_SLOWPATH = (1 << 0), /* In slow path processing */
|
|
|
|
CA_ACK_WIN_UPDATE = (1 << 1), /* ACK updated window */
|
|
|
|
CA_ACK_ECE = (1 << 2), /* ECE bit is set on ack */
|
2005-06-24 03:19:55 +08:00
|
|
|
};
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Interface for adding new TCP congestion control handlers
|
|
|
|
*/
|
|
|
|
#define TCP_CA_NAME_MAX 16
|
2006-11-10 08:32:06 +08:00
|
|
|
#define TCP_CA_MAX 128
|
|
|
|
#define TCP_CA_BUF_MAX (TCP_CA_NAME_MAX*TCP_CA_MAX)
|
|
|
|
|
net: tcp: add key management to congestion control
This patch adds necessary infrastructure to the congestion control
framework for later per route congestion control support.
For a per route congestion control possibility, our aim is to store
a unique u32 key identifier into dst metrics, which can then be
mapped into a tcp_congestion_ops struct. We argue that having a
RTAX key entry is the most simple, generic and easy way to manage,
and also keeps the memory footprint of dst entries lower on 64 bit
than with storing a pointer directly, for example. Having a unique
key id also allows for decoupling actual TCP congestion control
module management from the FIB layer, i.e. we don't have to care
about expensive module refcounting inside the FIB at this point.
We first thought of using an IDR store for the realization, which
takes over dynamic assignment of unused key space and also performs
the key to pointer mapping in RCU. While doing so, we stumbled upon
the issue that due to the nature of dynamic key distribution, it
just so happens, arguably in very rare occasions, that excessive
module loads and unloads can lead to a possible reuse of previously
used key space. Thus, previously stale keys in the dst metric are
now being reassigned to a different congestion control algorithm,
which might lead to unexpected behaviour. One way to resolve this
would have been to walk FIBs on the actually rare occasion of a
module unload and reset the metric keys for each FIB in each netns,
but that's just very costly.
Therefore, we argue a better solution is to reuse the unique
congestion control algorithm name member and map that into u32 key
space through jhash. For that, we split the flags attribute (as it
currently uses 2 bits only anyway) into two u32 attributes, flags
and key, so that we can keep the cacheline boundary of 2 cachelines
on x86_64 and cache the precalculated key at registration time for
the fast path. On average we might expect 2 - 4 modules being loaded
worst case perhaps 15, so a key collision possibility is extremely
low, and guaranteed collision-free on LE/BE for all in-tree modules.
Overall this results in much simpler code, and all without the
overhead of an IDR. Due to the deterministic nature, modules can
now be unloaded, the congestion control algorithm for a specific
but unloaded key will fall back to the default one, and on module
reload time it will switch back to the expected algorithm
transparently.
Joint work with Florian Westphal.
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:46 +08:00
|
|
|
#define TCP_CA_UNSPEC 0
|
|
|
|
|
2014-09-27 04:37:33 +08:00
|
|
|
/* Algorithm can be set on socket without CAP_NET_ADMIN privileges */
|
2007-04-24 13:26:16 +08:00
|
|
|
#define TCP_CONG_NON_RESTRICTED 0x1
|
2014-09-27 04:37:33 +08:00
|
|
|
/* Requires ECN/ECT set on all packets */
|
|
|
|
#define TCP_CONG_NEEDS_ECN 0x2
|
2007-04-24 13:26:16 +08:00
|
|
|
|
2015-04-29 07:23:48 +08:00
|
|
|
union tcp_cc_info;
|
|
|
|
|
2016-05-12 01:02:13 +08:00
|
|
|
struct ack_sample {
|
|
|
|
u32 pkts_acked;
|
|
|
|
s32 rtt_us;
|
2016-06-09 12:16:44 +08:00
|
|
|
u32 in_flight;
|
2016-05-12 01:02:13 +08:00
|
|
|
};
|
|
|
|
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
|
|
|
/* A rate sample measures the number of (original/retransmitted) data
|
|
|
|
* packets delivered "delivered" over an interval of time "interval_us".
|
|
|
|
* The tcp_rate.c code fills in the rate sample, and congestion
|
|
|
|
* control modules that define a cong_control function to run at the end
|
|
|
|
* of ACK processing can optionally chose to consult this sample when
|
|
|
|
* setting cwnd and pacing rate.
|
|
|
|
* A sample is invalid if "delivered" or "interval_us" is negative.
|
|
|
|
*/
|
|
|
|
struct rate_sample {
|
2017-05-17 05:00:14 +08:00
|
|
|
u64 prior_mstamp; /* starting timestamp for interval */
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
|
|
|
u32 prior_delivered; /* tp->delivered at "prior_mstamp" */
|
|
|
|
s32 delivered; /* number of packets delivered over interval */
|
|
|
|
long interval_us; /* time for tp->delivered to incr "delivered" */
|
|
|
|
long rtt_us; /* RTT of last (S)ACKed packet (or -1) */
|
|
|
|
int losses; /* number of packets marked lost upon ACK */
|
|
|
|
u32 acked_sacked; /* number of packets newly (S)ACKed upon ACK */
|
|
|
|
u32 prior_in_flight; /* in flight before this ACK */
|
2016-09-20 11:39:15 +08:00
|
|
|
bool is_app_limited; /* is sample from packet with bubble in pipe? */
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
|
|
|
bool is_retrans; /* is sample from retransmission? */
|
|
|
|
};
|
|
|
|
|
2005-06-24 03:19:55 +08:00
|
|
|
struct tcp_congestion_ops {
|
|
|
|
struct list_head list;
|
net: tcp: add key management to congestion control
This patch adds necessary infrastructure to the congestion control
framework for later per route congestion control support.
For a per route congestion control possibility, our aim is to store
a unique u32 key identifier into dst metrics, which can then be
mapped into a tcp_congestion_ops struct. We argue that having a
RTAX key entry is the most simple, generic and easy way to manage,
and also keeps the memory footprint of dst entries lower on 64 bit
than with storing a pointer directly, for example. Having a unique
key id also allows for decoupling actual TCP congestion control
module management from the FIB layer, i.e. we don't have to care
about expensive module refcounting inside the FIB at this point.
We first thought of using an IDR store for the realization, which
takes over dynamic assignment of unused key space and also performs
the key to pointer mapping in RCU. While doing so, we stumbled upon
the issue that due to the nature of dynamic key distribution, it
just so happens, arguably in very rare occasions, that excessive
module loads and unloads can lead to a possible reuse of previously
used key space. Thus, previously stale keys in the dst metric are
now being reassigned to a different congestion control algorithm,
which might lead to unexpected behaviour. One way to resolve this
would have been to walk FIBs on the actually rare occasion of a
module unload and reset the metric keys for each FIB in each netns,
but that's just very costly.
Therefore, we argue a better solution is to reuse the unique
congestion control algorithm name member and map that into u32 key
space through jhash. For that, we split the flags attribute (as it
currently uses 2 bits only anyway) into two u32 attributes, flags
and key, so that we can keep the cacheline boundary of 2 cachelines
on x86_64 and cache the precalculated key at registration time for
the fast path. On average we might expect 2 - 4 modules being loaded
worst case perhaps 15, so a key collision possibility is extremely
low, and guaranteed collision-free on LE/BE for all in-tree modules.
Overall this results in much simpler code, and all without the
overhead of an IDR. Due to the deterministic nature, modules can
now be unloaded, the congestion control algorithm for a specific
but unloaded key will fall back to the default one, and on module
reload time it will switch back to the expected algorithm
transparently.
Joint work with Florian Westphal.
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:46 +08:00
|
|
|
u32 key;
|
|
|
|
u32 flags;
|
2005-06-24 03:19:55 +08:00
|
|
|
|
|
|
|
/* initialize private data (optional) */
|
2005-08-10 15:03:31 +08:00
|
|
|
void (*init)(struct sock *sk);
|
2005-06-24 03:19:55 +08:00
|
|
|
/* cleanup private data (optional) */
|
2005-08-10 15:03:31 +08:00
|
|
|
void (*release)(struct sock *sk);
|
2005-06-24 03:19:55 +08:00
|
|
|
|
|
|
|
/* return slow start threshold (required) */
|
2005-08-10 15:03:31 +08:00
|
|
|
u32 (*ssthresh)(struct sock *sk);
|
2005-06-24 03:19:55 +08:00
|
|
|
/* do new cwnd calculation (required) */
|
2014-05-03 12:18:05 +08:00
|
|
|
void (*cong_avoid)(struct sock *sk, u32 ack, u32 acked);
|
2005-06-24 03:19:55 +08:00
|
|
|
/* call before changing ca_state (optional) */
|
2005-08-10 15:03:31 +08:00
|
|
|
void (*set_state)(struct sock *sk, u8 new_state);
|
2005-06-24 03:19:55 +08:00
|
|
|
/* call when cwnd event occurs (optional) */
|
2005-08-10 15:03:31 +08:00
|
|
|
void (*cwnd_event)(struct sock *sk, enum tcp_ca_event ev);
|
2014-09-27 04:37:34 +08:00
|
|
|
/* call when ack arrives (optional) */
|
|
|
|
void (*in_ack_event)(struct sock *sk, u32 flags);
|
2017-06-03 20:10:54 +08:00
|
|
|
/* new value of cwnd after loss (required) */
|
2005-08-10 15:03:31 +08:00
|
|
|
u32 (*undo_cwnd)(struct sock *sk);
|
2005-06-24 03:19:55 +08:00
|
|
|
/* hook for packet ack accounting (optional) */
|
2016-05-12 01:02:13 +08:00
|
|
|
void (*pkts_acked)(struct sock *sk, const struct ack_sample *sample);
|
2016-09-20 11:39:17 +08:00
|
|
|
/* suggest number of segments for each skb to transmit (optional) */
|
|
|
|
u32 (*tso_segs_goal)(struct sock *sk);
|
2016-09-20 11:39:20 +08:00
|
|
|
/* returns the multiplier used in tcp_sndbuf_expand (optional) */
|
|
|
|
u32 (*sndbuf_expand)(struct sock *sk);
|
2016-09-20 11:39:21 +08:00
|
|
|
/* call when packets are delivered to update cwnd and pacing rate,
|
|
|
|
* after all the ca_state processing. (optional)
|
|
|
|
*/
|
|
|
|
void (*cong_control)(struct sock *sk, const struct rate_sample *rs);
|
2005-08-12 23:51:49 +08:00
|
|
|
/* get info for inet_diag (optional) */
|
2015-04-29 07:23:48 +08:00
|
|
|
size_t (*get_info)(struct sock *sk, u32 ext, int *attr,
|
|
|
|
union tcp_cc_info *info);
|
2005-06-24 03:19:55 +08:00
|
|
|
|
|
|
|
char name[TCP_CA_NAME_MAX];
|
|
|
|
struct module *owner;
|
|
|
|
};
|
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_register_congestion_control(struct tcp_congestion_ops *type);
|
|
|
|
void tcp_unregister_congestion_control(struct tcp_congestion_ops *type);
|
2005-06-24 03:19:55 +08:00
|
|
|
|
2014-09-27 04:37:32 +08:00
|
|
|
void tcp_assign_congestion_control(struct sock *sk);
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_init_congestion_control(struct sock *sk);
|
|
|
|
void tcp_cleanup_congestion_control(struct sock *sk);
|
|
|
|
int tcp_set_default_congestion_control(const char *name);
|
|
|
|
void tcp_get_default_congestion_control(char *name);
|
|
|
|
void tcp_get_available_congestion_control(char *buf, size_t len);
|
|
|
|
void tcp_get_allowed_congestion_control(char *buf, size_t len);
|
|
|
|
int tcp_set_allowed_congestion_control(char *allowed);
|
|
|
|
int tcp_set_congestion_control(struct sock *sk, const char *name);
|
2015-01-29 09:01:35 +08:00
|
|
|
u32 tcp_slow_start(struct tcp_sock *tp, u32 acked);
|
|
|
|
void tcp_cong_avoid_ai(struct tcp_sock *tp, u32 w, u32 acked);
|
2005-06-24 03:19:55 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
u32 tcp_reno_ssthresh(struct sock *sk);
|
2016-11-21 21:18:38 +08:00
|
|
|
u32 tcp_reno_undo_cwnd(struct sock *sk);
|
2014-05-03 12:18:05 +08:00
|
|
|
void tcp_reno_cong_avoid(struct sock *sk, u32 ack, u32 acked);
|
2005-06-24 14:45:02 +08:00
|
|
|
extern struct tcp_congestion_ops tcp_reno;
|
2005-06-24 03:19:55 +08:00
|
|
|
|
net: tcp: add key management to congestion control
This patch adds necessary infrastructure to the congestion control
framework for later per route congestion control support.
For a per route congestion control possibility, our aim is to store
a unique u32 key identifier into dst metrics, which can then be
mapped into a tcp_congestion_ops struct. We argue that having a
RTAX key entry is the most simple, generic and easy way to manage,
and also keeps the memory footprint of dst entries lower on 64 bit
than with storing a pointer directly, for example. Having a unique
key id also allows for decoupling actual TCP congestion control
module management from the FIB layer, i.e. we don't have to care
about expensive module refcounting inside the FIB at this point.
We first thought of using an IDR store for the realization, which
takes over dynamic assignment of unused key space and also performs
the key to pointer mapping in RCU. While doing so, we stumbled upon
the issue that due to the nature of dynamic key distribution, it
just so happens, arguably in very rare occasions, that excessive
module loads and unloads can lead to a possible reuse of previously
used key space. Thus, previously stale keys in the dst metric are
now being reassigned to a different congestion control algorithm,
which might lead to unexpected behaviour. One way to resolve this
would have been to walk FIBs on the actually rare occasion of a
module unload and reset the metric keys for each FIB in each netns,
but that's just very costly.
Therefore, we argue a better solution is to reuse the unique
congestion control algorithm name member and map that into u32 key
space through jhash. For that, we split the flags attribute (as it
currently uses 2 bits only anyway) into two u32 attributes, flags
and key, so that we can keep the cacheline boundary of 2 cachelines
on x86_64 and cache the precalculated key at registration time for
the fast path. On average we might expect 2 - 4 modules being loaded
worst case perhaps 15, so a key collision possibility is extremely
low, and guaranteed collision-free on LE/BE for all in-tree modules.
Overall this results in much simpler code, and all without the
overhead of an IDR. Due to the deterministic nature, modules can
now be unloaded, the congestion control algorithm for a specific
but unloaded key will fall back to the default one, and on module
reload time it will switch back to the expected algorithm
transparently.
Joint work with Florian Westphal.
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:46 +08:00
|
|
|
struct tcp_congestion_ops *tcp_ca_find_key(u32 key);
|
tcp: use dctcp if enabled on the route to the initiator
Currently, the following case doesn't use DCTCP, even if it should:
A responder has f.e. Cubic as system wide default, but for a specific
route to the initiating host, DCTCP is being set in RTAX_CC_ALGO. The
initiating host then uses DCTCP as congestion control, but since the
initiator sets ECT(0), tcp_ecn_create_request() doesn't set ecn_ok,
and we have to fall back to Reno after 3WHS completes.
We were thinking on how to solve this in a minimal, non-intrusive
way without bloating tcp_ecn_create_request() needlessly: lets cache
the CA ecn option flag in RTAX_FEATURES. In other words, when ECT(0)
is set on the SYN packet, set ecn_ok=1 iff route RTAX_FEATURES
contains the unexposed (internal-only) DST_FEATURE_ECN_CA. This allows
to only do a single metric feature lookup inside tcp_ecn_create_request().
Joint work with Florian Westphal.
Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-08-31 21:58:47 +08:00
|
|
|
u32 tcp_ca_get_key_by_name(const char *name, bool *ecn_ca);
|
2015-01-06 06:57:47 +08:00
|
|
|
#ifdef CONFIG_INET
|
net: tcp: add key management to congestion control
This patch adds necessary infrastructure to the congestion control
framework for later per route congestion control support.
For a per route congestion control possibility, our aim is to store
a unique u32 key identifier into dst metrics, which can then be
mapped into a tcp_congestion_ops struct. We argue that having a
RTAX key entry is the most simple, generic and easy way to manage,
and also keeps the memory footprint of dst entries lower on 64 bit
than with storing a pointer directly, for example. Having a unique
key id also allows for decoupling actual TCP congestion control
module management from the FIB layer, i.e. we don't have to care
about expensive module refcounting inside the FIB at this point.
We first thought of using an IDR store for the realization, which
takes over dynamic assignment of unused key space and also performs
the key to pointer mapping in RCU. While doing so, we stumbled upon
the issue that due to the nature of dynamic key distribution, it
just so happens, arguably in very rare occasions, that excessive
module loads and unloads can lead to a possible reuse of previously
used key space. Thus, previously stale keys in the dst metric are
now being reassigned to a different congestion control algorithm,
which might lead to unexpected behaviour. One way to resolve this
would have been to walk FIBs on the actually rare occasion of a
module unload and reset the metric keys for each FIB in each netns,
but that's just very costly.
Therefore, we argue a better solution is to reuse the unique
congestion control algorithm name member and map that into u32 key
space through jhash. For that, we split the flags attribute (as it
currently uses 2 bits only anyway) into two u32 attributes, flags
and key, so that we can keep the cacheline boundary of 2 cachelines
on x86_64 and cache the precalculated key at registration time for
the fast path. On average we might expect 2 - 4 modules being loaded
worst case perhaps 15, so a key collision possibility is extremely
low, and guaranteed collision-free on LE/BE for all in-tree modules.
Overall this results in much simpler code, and all without the
overhead of an IDR. Due to the deterministic nature, modules can
now be unloaded, the congestion control algorithm for a specific
but unloaded key will fall back to the default one, and on module
reload time it will switch back to the expected algorithm
transparently.
Joint work with Florian Westphal.
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:46 +08:00
|
|
|
char *tcp_ca_get_name_by_key(u32 key, char *buffer);
|
2015-01-06 06:57:47 +08:00
|
|
|
#else
|
|
|
|
static inline char *tcp_ca_get_name_by_key(u32 key, char *buffer)
|
|
|
|
{
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
#endif
|
net: tcp: add key management to congestion control
This patch adds necessary infrastructure to the congestion control
framework for later per route congestion control support.
For a per route congestion control possibility, our aim is to store
a unique u32 key identifier into dst metrics, which can then be
mapped into a tcp_congestion_ops struct. We argue that having a
RTAX key entry is the most simple, generic and easy way to manage,
and also keeps the memory footprint of dst entries lower on 64 bit
than with storing a pointer directly, for example. Having a unique
key id also allows for decoupling actual TCP congestion control
module management from the FIB layer, i.e. we don't have to care
about expensive module refcounting inside the FIB at this point.
We first thought of using an IDR store for the realization, which
takes over dynamic assignment of unused key space and also performs
the key to pointer mapping in RCU. While doing so, we stumbled upon
the issue that due to the nature of dynamic key distribution, it
just so happens, arguably in very rare occasions, that excessive
module loads and unloads can lead to a possible reuse of previously
used key space. Thus, previously stale keys in the dst metric are
now being reassigned to a different congestion control algorithm,
which might lead to unexpected behaviour. One way to resolve this
would have been to walk FIBs on the actually rare occasion of a
module unload and reset the metric keys for each FIB in each netns,
but that's just very costly.
Therefore, we argue a better solution is to reuse the unique
congestion control algorithm name member and map that into u32 key
space through jhash. For that, we split the flags attribute (as it
currently uses 2 bits only anyway) into two u32 attributes, flags
and key, so that we can keep the cacheline boundary of 2 cachelines
on x86_64 and cache the precalculated key at registration time for
the fast path. On average we might expect 2 - 4 modules being loaded
worst case perhaps 15, so a key collision possibility is extremely
low, and guaranteed collision-free on LE/BE for all in-tree modules.
Overall this results in much simpler code, and all without the
overhead of an IDR. Due to the deterministic nature, modules can
now be unloaded, the congestion control algorithm for a specific
but unloaded key will fall back to the default one, and on module
reload time it will switch back to the expected algorithm
transparently.
Joint work with Florian Westphal.
Signed-off-by: Florian Westphal <fw@strlen.de>
Signed-off-by: Daniel Borkmann <dborkman@redhat.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:46 +08:00
|
|
|
|
2014-09-27 04:37:33 +08:00
|
|
|
static inline bool tcp_ca_needs_ecn(const struct sock *sk)
|
|
|
|
{
|
|
|
|
const struct inet_connection_sock *icsk = inet_csk(sk);
|
|
|
|
|
|
|
|
return icsk->icsk_ca_ops->flags & TCP_CONG_NEEDS_ECN;
|
|
|
|
}
|
|
|
|
|
2005-08-10 15:03:31 +08:00
|
|
|
static inline void tcp_set_ca_state(struct sock *sk, const u8 ca_state)
|
2005-06-24 03:19:55 +08:00
|
|
|
{
|
2005-08-10 15:03:31 +08:00
|
|
|
struct inet_connection_sock *icsk = inet_csk(sk);
|
|
|
|
|
|
|
|
if (icsk->icsk_ca_ops->set_state)
|
|
|
|
icsk->icsk_ca_ops->set_state(sk, ca_state);
|
|
|
|
icsk->icsk_ca_state = ca_state;
|
2005-06-24 03:19:55 +08:00
|
|
|
}
|
|
|
|
|
2005-08-10 15:03:31 +08:00
|
|
|
static inline void tcp_ca_event(struct sock *sk, const enum tcp_ca_event event)
|
2005-06-24 03:19:55 +08:00
|
|
|
{
|
2005-08-10 15:03:31 +08:00
|
|
|
const struct inet_connection_sock *icsk = inet_csk(sk);
|
|
|
|
|
|
|
|
if (icsk->icsk_ca_ops->cwnd_event)
|
|
|
|
icsk->icsk_ca_ops->cwnd_event(sk, event);
|
2005-06-24 03:19:55 +08:00
|
|
|
}
|
|
|
|
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
|
|
|
/* From tcp_rate.c */
|
|
|
|
void tcp_rate_skb_sent(struct sock *sk, struct sk_buff *skb);
|
|
|
|
void tcp_rate_skb_delivered(struct sock *sk, struct sk_buff *skb,
|
|
|
|
struct rate_sample *rs);
|
|
|
|
void tcp_rate_gen(struct sock *sk, u32 delivered, u32 lost,
|
2017-04-26 01:15:37 +08:00
|
|
|
struct rate_sample *rs);
|
2016-09-20 11:39:15 +08:00
|
|
|
void tcp_rate_check_app_limited(struct sock *sk);
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
|
|
|
|
2007-08-09 20:14:46 +08:00
|
|
|
/* These functions determine how the current flow behaves in respect of SACK
|
|
|
|
* handling. SACK is negotiated with the peer, and therefore it can vary
|
|
|
|
* between different flows.
|
|
|
|
*
|
|
|
|
* tcp_is_sack - SACK enabled
|
|
|
|
* tcp_is_reno - No SACK
|
|
|
|
* tcp_is_fack - FACK enabled, implies SACK enabled
|
|
|
|
*/
|
|
|
|
static inline int tcp_is_sack(const struct tcp_sock *tp)
|
|
|
|
{
|
|
|
|
return tp->rx_opt.sack_ok;
|
|
|
|
}
|
|
|
|
|
2012-05-17 07:15:34 +08:00
|
|
|
static inline bool tcp_is_reno(const struct tcp_sock *tp)
|
2007-08-09 20:14:46 +08:00
|
|
|
{
|
|
|
|
return !tcp_is_sack(tp);
|
|
|
|
}
|
|
|
|
|
2012-05-17 07:15:34 +08:00
|
|
|
static inline bool tcp_is_fack(const struct tcp_sock *tp)
|
2007-08-09 20:14:46 +08:00
|
|
|
{
|
2011-12-20 21:23:24 +08:00
|
|
|
return tp->rx_opt.sack_ok & TCP_FACK_ENABLED;
|
2007-08-09 20:14:46 +08:00
|
|
|
}
|
|
|
|
|
|
|
|
static inline void tcp_enable_fack(struct tcp_sock *tp)
|
|
|
|
{
|
2011-12-20 21:23:24 +08:00
|
|
|
tp->rx_opt.sack_ok |= TCP_FACK_ENABLED;
|
2007-08-09 20:14:46 +08:00
|
|
|
}
|
|
|
|
|
2007-08-09 19:37:30 +08:00
|
|
|
static inline unsigned int tcp_left_out(const struct tcp_sock *tp)
|
|
|
|
{
|
|
|
|
return tp->sacked_out + tp->lost_out;
|
|
|
|
}
|
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
/* This determines how many packets are "in the network" to the best
|
|
|
|
* of our knowledge. In many cases it is conservative, but where
|
|
|
|
* detailed information is available from the receiver (via SACK
|
|
|
|
* blocks etc.) we can make more aggressive calculations.
|
|
|
|
*
|
|
|
|
* Use this for decisions involving congestion control, use just
|
|
|
|
* tp->packets_out to determine if the send queue is empty or not.
|
|
|
|
*
|
|
|
|
* Read this equation as:
|
|
|
|
*
|
|
|
|
* "Packets sent once on transmission queue" MINUS
|
|
|
|
* "Packets left network, but not honestly ACKed yet" PLUS
|
|
|
|
* "Packets fast retransmitted"
|
|
|
|
*/
|
2006-01-04 08:03:49 +08:00
|
|
|
static inline unsigned int tcp_packets_in_flight(const struct tcp_sock *tp)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
2007-08-09 19:37:30 +08:00
|
|
|
return tp->packets_out - tcp_left_out(tp) + tp->retrans_out;
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
2009-09-15 16:30:10 +08:00
|
|
|
#define TCP_INFINITE_SSTHRESH 0x7fffffff
|
|
|
|
|
2015-07-10 04:16:29 +08:00
|
|
|
static inline bool tcp_in_slow_start(const struct tcp_sock *tp)
|
|
|
|
{
|
2015-07-10 04:16:30 +08:00
|
|
|
return tp->snd_cwnd < tp->snd_ssthresh;
|
2015-07-10 04:16:29 +08:00
|
|
|
}
|
|
|
|
|
2009-09-15 16:30:10 +08:00
|
|
|
static inline bool tcp_in_initial_slowstart(const struct tcp_sock *tp)
|
|
|
|
{
|
|
|
|
return tp->snd_ssthresh >= TCP_INFINITE_SSTHRESH;
|
|
|
|
}
|
|
|
|
|
2012-09-03 01:38:04 +08:00
|
|
|
static inline bool tcp_in_cwnd_reduction(const struct sock *sk)
|
|
|
|
{
|
|
|
|
return (TCPF_CA_CWR | TCPF_CA_Recovery) &
|
|
|
|
(1 << inet_csk(sk)->icsk_ca_state);
|
|
|
|
}
|
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
/* If cwnd > ssthresh, we may raise ssthresh to be half-way to cwnd.
|
2012-09-03 01:38:04 +08:00
|
|
|
* The exception is cwnd reduction phase, when cwnd is decreasing towards
|
2005-04-17 06:20:36 +08:00
|
|
|
* ssthresh.
|
|
|
|
*/
|
2005-08-10 15:03:31 +08:00
|
|
|
static inline __u32 tcp_current_ssthresh(const struct sock *sk)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
2005-08-10 15:03:31 +08:00
|
|
|
const struct tcp_sock *tp = tcp_sk(sk);
|
2011-10-21 17:22:42 +08:00
|
|
|
|
2012-09-03 01:38:04 +08:00
|
|
|
if (tcp_in_cwnd_reduction(sk))
|
2005-04-17 06:20:36 +08:00
|
|
|
return tp->snd_ssthresh;
|
|
|
|
else
|
|
|
|
return max(tp->snd_ssthresh,
|
|
|
|
((tp->snd_cwnd >> 1) +
|
|
|
|
(tp->snd_cwnd >> 2)));
|
|
|
|
}
|
|
|
|
|
2007-07-27 21:36:17 +08:00
|
|
|
/* Use define here intentionally to get WARN_ON location shown at the caller */
|
|
|
|
#define tcp_verify_left_out(tp) WARN_ON(tcp_left_out(tp) > tp->packets_out)
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2014-07-14 22:58:32 +08:00
|
|
|
void tcp_enter_cwr(struct sock *sk);
|
2013-09-24 02:33:32 +08:00
|
|
|
__u32 tcp_init_cwnd(const struct tcp_sock *tp, const struct dst_entry *dst);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2011-11-22 01:15:14 +08:00
|
|
|
/* The maximum number of MSS of available cwnd for which TSO defers
|
|
|
|
* sending if not using sysctl_tcp_tso_win_divisor.
|
|
|
|
*/
|
|
|
|
static inline __u32 tcp_max_tso_deferred_mss(const struct tcp_sock *tp)
|
|
|
|
{
|
|
|
|
return 3;
|
|
|
|
}
|
|
|
|
|
2007-12-31 20:48:41 +08:00
|
|
|
/* Returns end sequence number of the receiver's advertised window */
|
|
|
|
static inline u32 tcp_wnd_end(const struct tcp_sock *tp)
|
|
|
|
{
|
|
|
|
return tp->snd_una + tp->snd_wnd;
|
|
|
|
}
|
tcp: fix cwnd limited checking to improve congestion control
Yuchung discovered tcp_is_cwnd_limited() was returning false in
slow start phase even if the application filled the socket write queue.
All congestion modules take into account tcp_is_cwnd_limited()
before increasing cwnd, so this behavior limits slow start from
probing the bandwidth at full speed.
The problem is that even if write queue is full (aka we are _not_
application limited), cwnd can be under utilized if TSO should auto
defer or TCP Small queues decided to hold packets.
So the in_flight can be kept to smaller value, and we can get to the
point tcp_is_cwnd_limited() returns false.
With TCP Small Queues and FQ/pacing, this issue is more visible.
We fix this by having tcp_cwnd_validate(), which is supposed to track
such things, take into account unsent_segs, the number of segs that we
are not sending at the moment due to TSO or TSQ, but intend to send
real soon. Then when we are cwnd-limited, remember this fact while we
are processing the window of ACKs that comes back.
For example, suppose we have a brand new connection with cwnd=10; we
are in slow start, and we send a flight of 9 packets. By the time we
have received ACKs for all 9 packets we want our cwnd to be 18.
We implement this by setting tp->lsnd_pending to 9, and
considering ourselves to be cwnd-limited while cwnd is less than
twice tp->lsnd_pending (2*9 -> 18).
This makes tcp_is_cwnd_limited() more understandable, by removing
the GSO/TSO kludge, that tried to work around the issue.
Note the in_flight parameter can be removed in a followup cleanup
patch.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2014-05-01 02:58:13 +08:00
|
|
|
|
|
|
|
/* We follow the spirit of RFC2861 to validate cwnd but implement a more
|
|
|
|
* flexible approach. The RFC suggests cwnd should not be raised unless
|
2014-05-22 22:41:08 +08:00
|
|
|
* it was fully used previously. And that's exactly what we do in
|
|
|
|
* congestion avoidance mode. But in slow start we allow cwnd to grow
|
|
|
|
* as long as the application has used half the cwnd.
|
tcp: fix cwnd limited checking to improve congestion control
Yuchung discovered tcp_is_cwnd_limited() was returning false in
slow start phase even if the application filled the socket write queue.
All congestion modules take into account tcp_is_cwnd_limited()
before increasing cwnd, so this behavior limits slow start from
probing the bandwidth at full speed.
The problem is that even if write queue is full (aka we are _not_
application limited), cwnd can be under utilized if TSO should auto
defer or TCP Small queues decided to hold packets.
So the in_flight can be kept to smaller value, and we can get to the
point tcp_is_cwnd_limited() returns false.
With TCP Small Queues and FQ/pacing, this issue is more visible.
We fix this by having tcp_cwnd_validate(), which is supposed to track
such things, take into account unsent_segs, the number of segs that we
are not sending at the moment due to TSO or TSQ, but intend to send
real soon. Then when we are cwnd-limited, remember this fact while we
are processing the window of ACKs that comes back.
For example, suppose we have a brand new connection with cwnd=10; we
are in slow start, and we send a flight of 9 packets. By the time we
have received ACKs for all 9 packets we want our cwnd to be 18.
We implement this by setting tp->lsnd_pending to 9, and
considering ourselves to be cwnd-limited while cwnd is less than
twice tp->lsnd_pending (2*9 -> 18).
This makes tcp_is_cwnd_limited() more understandable, by removing
the GSO/TSO kludge, that tried to work around the issue.
Note the in_flight parameter can be removed in a followup cleanup
patch.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2014-05-01 02:58:13 +08:00
|
|
|
* Example :
|
|
|
|
* cwnd is 10 (IW10), but application sends 9 frames.
|
|
|
|
* We allow cwnd to reach 18 when all frames are ACKed.
|
|
|
|
* This check is safe because it's as aggressive as slow start which already
|
|
|
|
* risks 100% overshoot. The advantage is that we discourage application to
|
|
|
|
* either send more filler packets or data to artificially blow up the cwnd
|
|
|
|
* usage, and allow application-limited process to probe bw more aggressively.
|
|
|
|
*/
|
2014-05-03 12:18:05 +08:00
|
|
|
static inline bool tcp_is_cwnd_limited(const struct sock *sk)
|
tcp: fix cwnd limited checking to improve congestion control
Yuchung discovered tcp_is_cwnd_limited() was returning false in
slow start phase even if the application filled the socket write queue.
All congestion modules take into account tcp_is_cwnd_limited()
before increasing cwnd, so this behavior limits slow start from
probing the bandwidth at full speed.
The problem is that even if write queue is full (aka we are _not_
application limited), cwnd can be under utilized if TSO should auto
defer or TCP Small queues decided to hold packets.
So the in_flight can be kept to smaller value, and we can get to the
point tcp_is_cwnd_limited() returns false.
With TCP Small Queues and FQ/pacing, this issue is more visible.
We fix this by having tcp_cwnd_validate(), which is supposed to track
such things, take into account unsent_segs, the number of segs that we
are not sending at the moment due to TSO or TSQ, but intend to send
real soon. Then when we are cwnd-limited, remember this fact while we
are processing the window of ACKs that comes back.
For example, suppose we have a brand new connection with cwnd=10; we
are in slow start, and we send a flight of 9 packets. By the time we
have received ACKs for all 9 packets we want our cwnd to be 18.
We implement this by setting tp->lsnd_pending to 9, and
considering ourselves to be cwnd-limited while cwnd is less than
twice tp->lsnd_pending (2*9 -> 18).
This makes tcp_is_cwnd_limited() more understandable, by removing
the GSO/TSO kludge, that tried to work around the issue.
Note the in_flight parameter can be removed in a followup cleanup
patch.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2014-05-01 02:58:13 +08:00
|
|
|
{
|
|
|
|
const struct tcp_sock *tp = tcp_sk(sk);
|
|
|
|
|
2014-05-22 22:41:08 +08:00
|
|
|
/* If in slow start, ensure cwnd grows to twice what was ACKed. */
|
2015-07-10 04:16:29 +08:00
|
|
|
if (tcp_in_slow_start(tp))
|
2014-05-22 22:41:08 +08:00
|
|
|
return tp->snd_cwnd < 2 * tp->max_packets_out;
|
|
|
|
|
|
|
|
return tp->is_cwnd_limited;
|
tcp: fix cwnd limited checking to improve congestion control
Yuchung discovered tcp_is_cwnd_limited() was returning false in
slow start phase even if the application filled the socket write queue.
All congestion modules take into account tcp_is_cwnd_limited()
before increasing cwnd, so this behavior limits slow start from
probing the bandwidth at full speed.
The problem is that even if write queue is full (aka we are _not_
application limited), cwnd can be under utilized if TSO should auto
defer or TCP Small queues decided to hold packets.
So the in_flight can be kept to smaller value, and we can get to the
point tcp_is_cwnd_limited() returns false.
With TCP Small Queues and FQ/pacing, this issue is more visible.
We fix this by having tcp_cwnd_validate(), which is supposed to track
such things, take into account unsent_segs, the number of segs that we
are not sending at the moment due to TSO or TSQ, but intend to send
real soon. Then when we are cwnd-limited, remember this fact while we
are processing the window of ACKs that comes back.
For example, suppose we have a brand new connection with cwnd=10; we
are in slow start, and we send a flight of 9 packets. By the time we
have received ACKs for all 9 packets we want our cwnd to be 18.
We implement this by setting tp->lsnd_pending to 9, and
considering ourselves to be cwnd-limited while cwnd is less than
twice tp->lsnd_pending (2*9 -> 18).
This makes tcp_is_cwnd_limited() more understandable, by removing
the GSO/TSO kludge, that tried to work around the issue.
Note the in_flight parameter can be removed in a followup cleanup
patch.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2014-05-01 02:58:13 +08:00
|
|
|
}
|
2005-11-11 08:53:30 +08:00
|
|
|
|
tcp: adjust window probe timers to safer values
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>
2015-05-07 05:26:24 +08:00
|
|
|
/* Something is really bad, we could not queue an additional packet,
|
|
|
|
* because qdisc is full or receiver sent a 0 window.
|
|
|
|
* We do not want to add fuel to the fire, or abort too early,
|
|
|
|
* so make sure the timer we arm now is at least 200ms in the future,
|
|
|
|
* regardless of current icsk_rto value (as it could be ~2ms)
|
|
|
|
*/
|
|
|
|
static inline unsigned long tcp_probe0_base(const struct sock *sk)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
tcp: adjust window probe timers to safer values
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>
2015-05-07 05:26:24 +08:00
|
|
|
return max_t(unsigned long, inet_csk(sk)->icsk_rto, TCP_RTO_MIN);
|
|
|
|
}
|
[TCP]: Sed magic converts func(sk, tp, ...) -> func(sk, ...)
This is (mostly) automated change using magic:
sed -e '/struct sock \*sk/ N' -e '/struct sock \*sk/ N'
-e '/struct sock \*sk/ N' -e '/struct sock \*sk/ N'
-e 's|struct sock \*sk,[\n\t ]*struct tcp_sock \*tp\([^{]*\n{\n\)|
struct sock \*sk\1\tstruct tcp_sock *tp = tcp_sk(sk);\n|g'
-e 's|struct sock \*sk, struct tcp_sock \*tp|
struct sock \*sk|g' -e 's|sk, tp\([^-]\)|sk\1|g'
Fixed four unused variable (tp) warnings that were introduced.
In addition, manually added newlines after local variables and
tweaked function arguments positioning.
$ gcc --version
gcc (GCC) 4.1.1 20060525 (Red Hat 4.1.1-1)
...
$ codiff -fV built-in.o.old built-in.o.new
net/ipv4/route.c:
rt_cache_flush | +14
1 function changed, 14 bytes added
net/ipv4/tcp.c:
tcp_setsockopt | -5
tcp_sendpage | -25
tcp_sendmsg | -16
3 functions changed, 46 bytes removed
net/ipv4/tcp_input.c:
tcp_try_undo_recovery | +3
tcp_try_undo_dsack | +2
tcp_mark_head_lost | -12
tcp_ack | -15
tcp_event_data_recv | -32
tcp_rcv_state_process | -10
tcp_rcv_established | +1
7 functions changed, 6 bytes added, 69 bytes removed, diff: -63
net/ipv4/tcp_output.c:
update_send_head | -9
tcp_transmit_skb | +19
tcp_cwnd_validate | +1
tcp_write_wakeup | -17
__tcp_push_pending_frames | -25
tcp_push_one | -8
tcp_send_fin | -4
7 functions changed, 20 bytes added, 63 bytes removed, diff: -43
built-in.o.new:
18 functions changed, 40 bytes added, 178 bytes removed, diff: -138
Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
2007-04-21 13:18:02 +08:00
|
|
|
|
tcp: adjust window probe timers to safer values
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>
2015-05-07 05:26:24 +08:00
|
|
|
/* Variant of inet_csk_rto_backoff() used for zero window probes */
|
|
|
|
static inline unsigned long tcp_probe0_when(const struct sock *sk,
|
|
|
|
unsigned long max_when)
|
|
|
|
{
|
|
|
|
u64 when = (u64)tcp_probe0_base(sk) << inet_csk(sk)->icsk_backoff;
|
|
|
|
|
|
|
|
return (unsigned long)min_t(u64, when, max_when);
|
|
|
|
}
|
|
|
|
|
|
|
|
static inline void tcp_check_probe_timer(struct sock *sk)
|
|
|
|
{
|
|
|
|
if (!tcp_sk(sk)->packets_out && !inet_csk(sk)->icsk_pending)
|
2005-08-10 11:11:08 +08:00
|
|
|
inet_csk_reset_xmit_timer(sk, ICSK_TIME_PROBE0,
|
tcp: adjust window probe timers to safer values
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>
2015-05-07 05:26:24 +08:00
|
|
|
tcp_probe0_base(sk), TCP_RTO_MAX);
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
2009-03-03 14:42:02 +08:00
|
|
|
static inline void tcp_init_wl(struct tcp_sock *tp, u32 seq)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
|
|
|
tp->snd_wl1 = seq;
|
|
|
|
}
|
|
|
|
|
2009-03-03 14:42:02 +08:00
|
|
|
static inline void tcp_update_wl(struct tcp_sock *tp, u32 seq)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
|
|
|
tp->snd_wl1 = seq;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Calculate(/check) TCP checksum
|
|
|
|
*/
|
2007-02-05 12:15:27 +08:00
|
|
|
static inline __sum16 tcp_v4_check(int len, __be32 saddr,
|
|
|
|
__be32 daddr, __wsum base)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
|
|
|
return csum_tcpudp_magic(saddr,daddr,len,IPPROTO_TCP,base);
|
|
|
|
}
|
|
|
|
|
2006-11-15 13:40:42 +08:00
|
|
|
static inline __sum16 __tcp_checksum_complete(struct sk_buff *skb)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
2005-11-11 05:01:24 +08:00
|
|
|
return __skb_checksum_complete(skb);
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
2012-05-17 07:15:34 +08:00
|
|
|
static inline bool tcp_checksum_complete(struct sk_buff *skb)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
2007-04-10 02:59:39 +08:00
|
|
|
return !skb_csum_unnecessary(skb) &&
|
2005-04-17 06:20:36 +08:00
|
|
|
__tcp_checksum_complete(skb);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Prequeue for VJ style copy to user, combined with checksumming. */
|
|
|
|
|
2006-01-04 08:03:49 +08:00
|
|
|
static inline void tcp_prequeue_init(struct tcp_sock *tp)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
|
|
|
tp->ucopy.task = NULL;
|
|
|
|
tp->ucopy.len = 0;
|
|
|
|
tp->ucopy.memory = 0;
|
|
|
|
skb_queue_head_init(&tp->ucopy.prequeue);
|
|
|
|
}
|
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
bool tcp_prequeue(struct sock *sk, struct sk_buff *skb);
|
2016-08-27 22:37:54 +08:00
|
|
|
bool tcp_add_backlog(struct sock *sk, struct sk_buff *skb);
|
2016-11-11 05:12:35 +08:00
|
|
|
int tcp_filter(struct sock *sk, struct sk_buff *skb);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
|
|
|
#undef STATE_TRACE
|
|
|
|
|
|
|
|
#ifdef STATE_TRACE
|
|
|
|
static const char *statename[]={
|
|
|
|
"Unused","Established","Syn Sent","Syn Recv",
|
|
|
|
"Fin Wait 1","Fin Wait 2","Time Wait", "Close",
|
|
|
|
"Close Wait","Last ACK","Listen","Closing"
|
|
|
|
};
|
|
|
|
#endif
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_set_state(struct sock *sk, int state);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_done(struct sock *sk);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2015-12-16 11:30:05 +08:00
|
|
|
int tcp_abort(struct sock *sk, int err);
|
|
|
|
|
2006-01-04 08:03:49 +08:00
|
|
|
static inline void tcp_sack_reset(struct tcp_options_received *rx_opt)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
|
|
|
rx_opt->dsack = 0;
|
|
|
|
rx_opt->num_sacks = 0;
|
|
|
|
}
|
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
u32 tcp_default_init_rwnd(u32 mss);
|
tcp: fix slow start after idle vs TSO/GSO
slow start after idle might reduce cwnd, but we perform this
after first packet was cooked and sent.
With TSO/GSO, it means that we might send a full TSO packet
even if cwnd should have been reduced to IW10.
Moving the SSAI check in skb_entail() makes sense, because
we slightly reduce number of times this check is done,
especially for large send() and TCP Small queue callbacks from
softirq context.
As Neal pointed out, we also need to perform the check
if/when receive window opens.
Tested:
Following packetdrill test demonstrates the problem
// Test of slow start after idle
`sysctl -q net.ipv4.tcp_slow_start_after_idle=1`
0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3
+0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0
+0 bind(3, ..., ...) = 0
+0 listen(3, 1) = 0
+0 < S 0:0(0) win 65535 <mss 1000,sackOK,nop,nop,nop,wscale 7>
+0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6>
+.100 < . 1:1(0) ack 1 win 511
+0 accept(3, ..., ...) = 4
+0 setsockopt(4, SOL_SOCKET, SO_SNDBUF, [200000], 4) = 0
+0 write(4, ..., 26000) = 26000
+0 > . 1:5001(5000) ack 1
+0 > . 5001:10001(5000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10 }%
+.100 < . 1:1(0) ack 10001 win 511
+0 %{ assert tcpi_snd_cwnd == 20, tcpi_snd_cwnd }%
+0 > . 10001:20001(10000) ack 1
+0 > P. 20001:26001(6000) ack 1
+.100 < . 1:1(0) ack 26001 win 511
+0 %{ assert tcpi_snd_cwnd == 36, tcpi_snd_cwnd }%
+4 write(4, ..., 20000) = 20000
// If slow start after idle works properly, we should send 5 MSS here (cwnd/2)
+0 > . 26001:31001(5000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }%
+0 > . 31001:36001(5000) ack 1
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-08-22 03:30:00 +08:00
|
|
|
void tcp_cwnd_restart(struct sock *sk, s32 delta);
|
|
|
|
|
|
|
|
static inline void tcp_slow_start_after_idle_check(struct sock *sk)
|
|
|
|
{
|
2017-05-06 03:53:23 +08:00
|
|
|
const struct tcp_congestion_ops *ca_ops = inet_csk(sk)->icsk_ca_ops;
|
tcp: fix slow start after idle vs TSO/GSO
slow start after idle might reduce cwnd, but we perform this
after first packet was cooked and sent.
With TSO/GSO, it means that we might send a full TSO packet
even if cwnd should have been reduced to IW10.
Moving the SSAI check in skb_entail() makes sense, because
we slightly reduce number of times this check is done,
especially for large send() and TCP Small queue callbacks from
softirq context.
As Neal pointed out, we also need to perform the check
if/when receive window opens.
Tested:
Following packetdrill test demonstrates the problem
// Test of slow start after idle
`sysctl -q net.ipv4.tcp_slow_start_after_idle=1`
0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3
+0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0
+0 bind(3, ..., ...) = 0
+0 listen(3, 1) = 0
+0 < S 0:0(0) win 65535 <mss 1000,sackOK,nop,nop,nop,wscale 7>
+0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6>
+.100 < . 1:1(0) ack 1 win 511
+0 accept(3, ..., ...) = 4
+0 setsockopt(4, SOL_SOCKET, SO_SNDBUF, [200000], 4) = 0
+0 write(4, ..., 26000) = 26000
+0 > . 1:5001(5000) ack 1
+0 > . 5001:10001(5000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10 }%
+.100 < . 1:1(0) ack 10001 win 511
+0 %{ assert tcpi_snd_cwnd == 20, tcpi_snd_cwnd }%
+0 > . 10001:20001(10000) ack 1
+0 > P. 20001:26001(6000) ack 1
+.100 < . 1:1(0) ack 26001 win 511
+0 %{ assert tcpi_snd_cwnd == 36, tcpi_snd_cwnd }%
+4 write(4, ..., 20000) = 20000
// If slow start after idle works properly, we should send 5 MSS here (cwnd/2)
+0 > . 26001:31001(5000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }%
+0 > . 31001:36001(5000) ack 1
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-08-22 03:30:00 +08:00
|
|
|
struct tcp_sock *tp = tcp_sk(sk);
|
|
|
|
s32 delta;
|
|
|
|
|
2017-05-06 03:53:23 +08:00
|
|
|
if (!sysctl_tcp_slow_start_after_idle || tp->packets_out ||
|
|
|
|
ca_ops->cong_control)
|
tcp: fix slow start after idle vs TSO/GSO
slow start after idle might reduce cwnd, but we perform this
after first packet was cooked and sent.
With TSO/GSO, it means that we might send a full TSO packet
even if cwnd should have been reduced to IW10.
Moving the SSAI check in skb_entail() makes sense, because
we slightly reduce number of times this check is done,
especially for large send() and TCP Small queue callbacks from
softirq context.
As Neal pointed out, we also need to perform the check
if/when receive window opens.
Tested:
Following packetdrill test demonstrates the problem
// Test of slow start after idle
`sysctl -q net.ipv4.tcp_slow_start_after_idle=1`
0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3
+0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0
+0 bind(3, ..., ...) = 0
+0 listen(3, 1) = 0
+0 < S 0:0(0) win 65535 <mss 1000,sackOK,nop,nop,nop,wscale 7>
+0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6>
+.100 < . 1:1(0) ack 1 win 511
+0 accept(3, ..., ...) = 4
+0 setsockopt(4, SOL_SOCKET, SO_SNDBUF, [200000], 4) = 0
+0 write(4, ..., 26000) = 26000
+0 > . 1:5001(5000) ack 1
+0 > . 5001:10001(5000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10 }%
+.100 < . 1:1(0) ack 10001 win 511
+0 %{ assert tcpi_snd_cwnd == 20, tcpi_snd_cwnd }%
+0 > . 10001:20001(10000) ack 1
+0 > P. 20001:26001(6000) ack 1
+.100 < . 1:1(0) ack 26001 win 511
+0 %{ assert tcpi_snd_cwnd == 36, tcpi_snd_cwnd }%
+4 write(4, ..., 20000) = 20000
// If slow start after idle works properly, we should send 5 MSS here (cwnd/2)
+0 > . 26001:31001(5000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }%
+0 > . 31001:36001(5000) ack 1
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-08-22 03:30:00 +08:00
|
|
|
return;
|
2017-05-17 05:00:03 +08:00
|
|
|
delta = tcp_jiffies32 - tp->lsndtime;
|
tcp: fix slow start after idle vs TSO/GSO
slow start after idle might reduce cwnd, but we perform this
after first packet was cooked and sent.
With TSO/GSO, it means that we might send a full TSO packet
even if cwnd should have been reduced to IW10.
Moving the SSAI check in skb_entail() makes sense, because
we slightly reduce number of times this check is done,
especially for large send() and TCP Small queue callbacks from
softirq context.
As Neal pointed out, we also need to perform the check
if/when receive window opens.
Tested:
Following packetdrill test demonstrates the problem
// Test of slow start after idle
`sysctl -q net.ipv4.tcp_slow_start_after_idle=1`
0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3
+0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0
+0 bind(3, ..., ...) = 0
+0 listen(3, 1) = 0
+0 < S 0:0(0) win 65535 <mss 1000,sackOK,nop,nop,nop,wscale 7>
+0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6>
+.100 < . 1:1(0) ack 1 win 511
+0 accept(3, ..., ...) = 4
+0 setsockopt(4, SOL_SOCKET, SO_SNDBUF, [200000], 4) = 0
+0 write(4, ..., 26000) = 26000
+0 > . 1:5001(5000) ack 1
+0 > . 5001:10001(5000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10 }%
+.100 < . 1:1(0) ack 10001 win 511
+0 %{ assert tcpi_snd_cwnd == 20, tcpi_snd_cwnd }%
+0 > . 10001:20001(10000) ack 1
+0 > P. 20001:26001(6000) ack 1
+.100 < . 1:1(0) ack 26001 win 511
+0 %{ assert tcpi_snd_cwnd == 36, tcpi_snd_cwnd }%
+4 write(4, ..., 20000) = 20000
// If slow start after idle works properly, we should send 5 MSS here (cwnd/2)
+0 > . 26001:31001(5000) ack 1
+0 %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }%
+0 > . 31001:36001(5000) ack 1
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-08-22 03:30:00 +08:00
|
|
|
if (delta > inet_csk(sk)->icsk_rto)
|
|
|
|
tcp_cwnd_restart(sk, delta);
|
|
|
|
}
|
2013-06-12 06:35:32 +08:00
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
/* Determine a window scaling and initial window to offer. */
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_select_initial_window(int __space, __u32 mss, __u32 *rcv_wnd,
|
|
|
|
__u32 *window_clamp, int wscale_ok,
|
|
|
|
__u8 *rcv_wscale, __u32 init_rcv_wnd);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
|
|
|
static inline int tcp_win_from_space(int space)
|
|
|
|
{
|
2017-03-24 07:05:12 +08:00
|
|
|
int tcp_adv_win_scale = sysctl_tcp_adv_win_scale;
|
|
|
|
|
|
|
|
return tcp_adv_win_scale <= 0 ?
|
|
|
|
(space>>(-tcp_adv_win_scale)) :
|
|
|
|
space - (space>>tcp_adv_win_scale);
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
2014-10-20 17:15:50 +08:00
|
|
|
/* Note: caller must be prepared to deal with negative returns */
|
2005-04-17 06:20:36 +08:00
|
|
|
static inline int tcp_space(const struct sock *sk)
|
|
|
|
{
|
|
|
|
return tcp_win_from_space(sk->sk_rcvbuf -
|
|
|
|
atomic_read(&sk->sk_rmem_alloc));
|
2014-10-20 17:15:50 +08:00
|
|
|
}
|
2005-04-17 06:20:36 +08:00
|
|
|
|
|
|
|
static inline int tcp_full_space(const struct sock *sk)
|
|
|
|
{
|
2014-10-20 17:15:50 +08:00
|
|
|
return tcp_win_from_space(sk->sk_rcvbuf);
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
2014-05-12 11:22:11 +08:00
|
|
|
extern void tcp_openreq_init_rwin(struct request_sock *req,
|
2015-09-25 22:39:09 +08:00
|
|
|
const struct sock *sk_listener,
|
|
|
|
const struct dst_entry *dst);
|
2014-05-12 11:22:11 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_enter_memory_pressure(struct sock *sk);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
|
|
|
static inline int keepalive_intvl_when(const struct tcp_sock *tp)
|
|
|
|
{
|
2016-01-07 22:38:45 +08:00
|
|
|
struct net *net = sock_net((struct sock *)tp);
|
|
|
|
|
|
|
|
return tp->keepalive_intvl ? : net->ipv4.sysctl_tcp_keepalive_intvl;
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
|
|
|
static inline int keepalive_time_when(const struct tcp_sock *tp)
|
|
|
|
{
|
2016-01-07 22:38:43 +08:00
|
|
|
struct net *net = sock_net((struct sock *)tp);
|
|
|
|
|
|
|
|
return tp->keepalive_time ? : net->ipv4.sysctl_tcp_keepalive_time;
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
2009-08-29 14:48:54 +08:00
|
|
|
static inline int keepalive_probes(const struct tcp_sock *tp)
|
|
|
|
{
|
2016-01-07 22:38:44 +08:00
|
|
|
struct net *net = sock_net((struct sock *)tp);
|
|
|
|
|
|
|
|
return tp->keepalive_probes ? : net->ipv4.sysctl_tcp_keepalive_probes;
|
2009-08-29 14:48:54 +08:00
|
|
|
}
|
|
|
|
|
2010-04-27 02:33:27 +08:00
|
|
|
static inline u32 keepalive_time_elapsed(const struct tcp_sock *tp)
|
|
|
|
{
|
|
|
|
const struct inet_connection_sock *icsk = &tp->inet_conn;
|
|
|
|
|
2017-05-17 05:00:07 +08:00
|
|
|
return min_t(u32, tcp_jiffies32 - icsk->icsk_ack.lrcvtime,
|
|
|
|
tcp_jiffies32 - tp->rcv_tstamp);
|
2010-04-27 02:33:27 +08:00
|
|
|
}
|
|
|
|
|
2005-08-10 11:10:42 +08:00
|
|
|
static inline int tcp_fin_time(const struct sock *sk)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
2016-02-03 15:46:56 +08:00
|
|
|
int fin_timeout = tcp_sk(sk)->linger2 ? : sock_net(sk)->ipv4.sysctl_tcp_fin_timeout;
|
2005-08-10 11:10:42 +08:00
|
|
|
const int rto = inet_csk(sk)->icsk_rto;
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2005-08-10 11:10:42 +08:00
|
|
|
if (fin_timeout < (rto << 2) - (rto >> 1))
|
|
|
|
fin_timeout = (rto << 2) - (rto >> 1);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
|
|
|
return fin_timeout;
|
|
|
|
}
|
|
|
|
|
2012-05-17 07:15:34 +08:00
|
|
|
static inline bool tcp_paws_check(const struct tcp_options_received *rx_opt,
|
|
|
|
int paws_win)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
2009-03-14 22:23:03 +08:00
|
|
|
if ((s32)(rx_opt->ts_recent - rx_opt->rcv_tsval) <= paws_win)
|
2012-05-17 07:15:34 +08:00
|
|
|
return true;
|
2009-03-14 22:23:03 +08:00
|
|
|
if (unlikely(get_seconds() >= rx_opt->ts_recent_stamp + TCP_PAWS_24DAYS))
|
2012-05-17 07:15:34 +08:00
|
|
|
return true;
|
2010-12-17 06:08:34 +08:00
|
|
|
/*
|
|
|
|
* Some OSes send SYN and SYNACK messages with tsval=0 tsecr=0,
|
|
|
|
* then following tcp messages have valid values. Ignore 0 value,
|
|
|
|
* or else 'negative' tsval might forbid us to accept their packets.
|
|
|
|
*/
|
|
|
|
if (!rx_opt->ts_recent)
|
2012-05-17 07:15:34 +08:00
|
|
|
return true;
|
|
|
|
return false;
|
2009-03-14 22:23:03 +08:00
|
|
|
}
|
|
|
|
|
2012-05-17 07:15:34 +08:00
|
|
|
static inline bool tcp_paws_reject(const struct tcp_options_received *rx_opt,
|
|
|
|
int rst)
|
2009-03-14 22:23:03 +08:00
|
|
|
{
|
|
|
|
if (tcp_paws_check(rx_opt, 0))
|
2012-05-17 07:15:34 +08:00
|
|
|
return false;
|
2005-04-17 06:20:36 +08:00
|
|
|
|
|
|
|
/* RST segments are not recommended to carry timestamp,
|
|
|
|
and, if they do, it is recommended to ignore PAWS because
|
|
|
|
"their cleanup function should take precedence over timestamps."
|
|
|
|
Certainly, it is mistake. It is necessary to understand the reasons
|
|
|
|
of this constraint to relax it: if peer reboots, clock may go
|
|
|
|
out-of-sync and half-open connections will not be reset.
|
|
|
|
Actually, the problem would be not existing if all
|
|
|
|
the implementations followed draft about maintaining clock
|
|
|
|
via reboots. Linux-2.2 DOES NOT!
|
|
|
|
|
|
|
|
However, we can relax time bounds for RST segments to MSL.
|
|
|
|
*/
|
2007-03-05 08:12:44 +08:00
|
|
|
if (rst && get_seconds() >= rx_opt->ts_recent_stamp + TCP_PAWS_MSL)
|
2012-05-17 07:15:34 +08:00
|
|
|
return false;
|
|
|
|
return true;
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
2015-03-17 12:06:20 +08:00
|
|
|
bool tcp_oow_rate_limited(struct net *net, const struct sk_buff *skb,
|
|
|
|
int mib_idx, u32 *last_oow_ack_time);
|
tcp: helpers to mitigate ACK loops by rate-limiting out-of-window dupacks
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>
2015-02-07 05:04:38 +08:00
|
|
|
|
2008-07-17 11:21:42 +08:00
|
|
|
static inline void tcp_mib_init(struct net *net)
|
2005-04-17 06:20:36 +08:00
|
|
|
{
|
|
|
|
/* See RFC 2012 */
|
net: snmp: kill various STATS_USER() helpers
In the old days (before linux-3.0), SNMP counters were duplicated,
one for user context, and one for BH context.
After commit 8f0ea0fe3a03 ("snmp: reduce percpu needs by 50%")
we have a single copy, and what really matters is preemption being
enabled or disabled, since we use this_cpu_inc() or __this_cpu_inc()
respectively.
We therefore kill SNMP_INC_STATS_USER(), SNMP_ADD_STATS_USER(),
NET_INC_STATS_USER(), NET_ADD_STATS_USER(), SCTP_INC_STATS_USER(),
SNMP_INC_STATS64_USER(), SNMP_ADD_STATS64_USER(), TCP_ADD_STATS_USER(),
UDP_INC_STATS_USER(), UDP6_INC_STATS_USER(), and XFRM_INC_STATS_USER()
Following patches will rename __BH helpers to make clear their
usage is not tied to BH being disabled.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-04-28 07:44:27 +08:00
|
|
|
TCP_ADD_STATS(net, TCP_MIB_RTOALGORITHM, 1);
|
|
|
|
TCP_ADD_STATS(net, TCP_MIB_RTOMIN, TCP_RTO_MIN*1000/HZ);
|
|
|
|
TCP_ADD_STATS(net, TCP_MIB_RTOMAX, TCP_RTO_MAX*1000/HZ);
|
|
|
|
TCP_ADD_STATS(net, TCP_MIB_MAXCONN, -1);
|
2005-04-17 06:20:36 +08:00
|
|
|
}
|
|
|
|
|
2007-09-21 02:30:48 +08:00
|
|
|
/* from STCP */
|
2008-09-21 12:25:15 +08:00
|
|
|
static inline void tcp_clear_retrans_hints_partial(struct tcp_sock *tp)
|
2007-09-21 02:40:37 +08:00
|
|
|
{
|
2005-11-11 09:14:59 +08:00
|
|
|
tp->lost_skb_hint = NULL;
|
2008-09-21 12:25:15 +08:00
|
|
|
}
|
|
|
|
|
|
|
|
static inline void tcp_clear_all_retrans_hints(struct tcp_sock *tp)
|
|
|
|
{
|
|
|
|
tcp_clear_retrans_hints_partial(tp);
|
2005-11-11 09:14:59 +08:00
|
|
|
tp->retransmit_skb_hint = NULL;
|
2007-09-21 02:37:19 +08:00
|
|
|
}
|
|
|
|
|
2012-01-31 13:18:33 +08:00
|
|
|
union tcp_md5_addr {
|
|
|
|
struct in_addr a4;
|
|
|
|
#if IS_ENABLED(CONFIG_IPV6)
|
|
|
|
struct in6_addr a6;
|
|
|
|
#endif
|
|
|
|
};
|
|
|
|
|
2006-11-15 11:07:45 +08:00
|
|
|
/* - key database */
|
|
|
|
struct tcp_md5sig_key {
|
2012-01-31 13:18:33 +08:00
|
|
|
struct hlist_node node;
|
2006-11-15 11:07:45 +08:00
|
|
|
u8 keylen;
|
2012-01-31 13:18:33 +08:00
|
|
|
u8 family; /* AF_INET or AF_INET6 */
|
|
|
|
union tcp_md5_addr addr;
|
|
|
|
u8 key[TCP_MD5SIG_MAXKEYLEN];
|
|
|
|
struct rcu_head rcu;
|
2006-11-15 11:07:45 +08:00
|
|
|
};
|
|
|
|
|
|
|
|
/* - sock block */
|
|
|
|
struct tcp_md5sig_info {
|
2012-01-31 13:18:33 +08:00
|
|
|
struct hlist_head head;
|
2012-02-01 02:45:40 +08:00
|
|
|
struct rcu_head rcu;
|
2006-11-15 11:07:45 +08:00
|
|
|
};
|
|
|
|
|
|
|
|
/* - pseudo header */
|
|
|
|
struct tcp4_pseudohdr {
|
|
|
|
__be32 saddr;
|
|
|
|
__be32 daddr;
|
|
|
|
__u8 pad;
|
|
|
|
__u8 protocol;
|
|
|
|
__be16 len;
|
|
|
|
};
|
|
|
|
|
|
|
|
struct tcp6_pseudohdr {
|
|
|
|
struct in6_addr saddr;
|
|
|
|
struct in6_addr daddr;
|
|
|
|
__be32 len;
|
|
|
|
__be32 protocol; /* including padding */
|
|
|
|
};
|
|
|
|
|
|
|
|
union tcp_md5sum_block {
|
|
|
|
struct tcp4_pseudohdr ip4;
|
2011-12-10 17:48:31 +08:00
|
|
|
#if IS_ENABLED(CONFIG_IPV6)
|
2006-11-15 11:07:45 +08:00
|
|
|
struct tcp6_pseudohdr ip6;
|
|
|
|
#endif
|
|
|
|
};
|
|
|
|
|
|
|
|
/* - pool: digest algorithm, hash description and scratch buffer */
|
|
|
|
struct tcp_md5sig_pool {
|
2016-01-24 21:20:23 +08:00
|
|
|
struct ahash_request *md5_req;
|
2016-06-28 00:51:53 +08:00
|
|
|
void *scratch;
|
2006-11-15 11:07:45 +08:00
|
|
|
};
|
|
|
|
|
|
|
|
/* - functions */
|
2015-03-25 06:58:55 +08:00
|
|
|
int tcp_v4_md5_hash_skb(char *md5_hash, const struct tcp_md5sig_key *key,
|
|
|
|
const struct sock *sk, const struct sk_buff *skb);
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_md5_do_add(struct sock *sk, const union tcp_md5_addr *addr,
|
|
|
|
int family, const u8 *newkey, u8 newkeylen, gfp_t gfp);
|
|
|
|
int tcp_md5_do_del(struct sock *sk, const union tcp_md5_addr *addr,
|
|
|
|
int family);
|
2015-09-25 22:39:15 +08:00
|
|
|
struct tcp_md5sig_key *tcp_v4_md5_lookup(const struct sock *sk,
|
2015-03-25 06:58:56 +08:00
|
|
|
const struct sock *addr_sk);
|
2006-11-15 11:07:45 +08:00
|
|
|
|
2008-04-18 11:45:16 +08:00
|
|
|
#ifdef CONFIG_TCP_MD5SIG
|
2015-09-25 22:39:15 +08:00
|
|
|
struct tcp_md5sig_key *tcp_md5_do_lookup(const struct sock *sk,
|
2013-09-24 02:33:32 +08:00
|
|
|
const union tcp_md5_addr *addr,
|
|
|
|
int family);
|
2012-01-31 13:18:33 +08:00
|
|
|
#define tcp_twsk_md5_key(twsk) ((twsk)->tw_md5_key)
|
2008-04-18 11:45:16 +08:00
|
|
|
#else
|
2015-09-25 22:39:15 +08:00
|
|
|
static inline struct tcp_md5sig_key *tcp_md5_do_lookup(const struct sock *sk,
|
2012-01-31 13:18:33 +08:00
|
|
|
const union tcp_md5_addr *addr,
|
|
|
|
int family)
|
|
|
|
{
|
|
|
|
return NULL;
|
|
|
|
}
|
2008-04-18 11:45:16 +08:00
|
|
|
#define tcp_twsk_md5_key(twsk) NULL
|
|
|
|
#endif
|
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
bool tcp_alloc_md5sig_pool(void);
|
2006-11-15 11:07:45 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
struct tcp_md5sig_pool *tcp_get_md5sig_pool(void);
|
2013-05-20 14:52:26 +08:00
|
|
|
static inline void tcp_put_md5sig_pool(void)
|
|
|
|
{
|
|
|
|
local_bh_enable();
|
|
|
|
}
|
2010-05-16 15:34:04 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_md5_hash_skb_data(struct tcp_md5sig_pool *, const struct sk_buff *,
|
|
|
|
unsigned int header_len);
|
|
|
|
int tcp_md5_hash_key(struct tcp_md5sig_pool *hp,
|
|
|
|
const struct tcp_md5sig_key *key);
|
2006-11-15 11:07:45 +08:00
|
|
|
|
2012-08-31 20:29:11 +08:00
|
|
|
/* From tcp_fastopen.c */
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_fastopen_cache_get(struct sock *sk, u16 *mss,
|
|
|
|
struct tcp_fastopen_cookie *cookie, int *syn_loss,
|
|
|
|
unsigned long *last_syn_loss);
|
|
|
|
void tcp_fastopen_cache_set(struct sock *sk, u16 mss,
|
2015-04-07 05:37:27 +08:00
|
|
|
struct tcp_fastopen_cookie *cookie, bool syn_lost,
|
|
|
|
u16 try_exp);
|
2012-07-19 14:43:07 +08:00
|
|
|
struct tcp_fastopen_request {
|
|
|
|
/* Fast Open cookie. Size 0 means a cookie request */
|
|
|
|
struct tcp_fastopen_cookie cookie;
|
|
|
|
struct msghdr *data; /* data in MSG_FASTOPEN */
|
2014-02-21 02:09:18 +08:00
|
|
|
size_t size;
|
|
|
|
int copied; /* queued in tcp_connect() */
|
2012-07-19 14:43:07 +08:00
|
|
|
};
|
|
|
|
void tcp_free_fastopen_req(struct tcp_sock *tp);
|
|
|
|
|
2012-08-31 20:29:11 +08:00
|
|
|
extern struct tcp_fastopen_context __rcu *tcp_fastopen_ctx;
|
|
|
|
int tcp_fastopen_reset_cipher(void *key, unsigned int len);
|
2016-02-02 13:03:07 +08:00
|
|
|
void tcp_fastopen_add_skb(struct sock *sk, struct sk_buff *skb);
|
2015-09-25 08:16:05 +08:00
|
|
|
struct sock *tcp_try_fastopen(struct sock *sk, struct sk_buff *skb,
|
|
|
|
struct request_sock *req,
|
|
|
|
struct tcp_fastopen_cookie *foc,
|
|
|
|
struct dst_entry *dst);
|
2013-10-20 03:48:58 +08:00
|
|
|
void tcp_fastopen_init_key_once(bool publish);
|
2017-01-24 02:59:20 +08:00
|
|
|
bool tcp_fastopen_cookie_check(struct sock *sk, u16 *mss,
|
|
|
|
struct tcp_fastopen_cookie *cookie);
|
net/tcp-fastopen: Add new API support
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: Wei Wang <weiwan@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2017-01-24 02:59:22 +08:00
|
|
|
bool tcp_fastopen_defer_connect(struct sock *sk, int *err);
|
2012-08-31 20:29:11 +08:00
|
|
|
#define TCP_FASTOPEN_KEY_LENGTH 16
|
|
|
|
|
|
|
|
/* Fastopen key context */
|
|
|
|
struct tcp_fastopen_context {
|
2013-06-25 16:21:06 +08:00
|
|
|
struct crypto_cipher *tfm;
|
|
|
|
__u8 key[TCP_FASTOPEN_KEY_LENGTH];
|
|
|
|
struct rcu_head rcu;
|
2012-08-31 20:29:11 +08:00
|
|
|
};
|
|
|
|
|
net/tcp_fastopen: Disable active side TFO in certain scenarios
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: Wei Wang <weiwan@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2017-04-21 05:45:46 +08:00
|
|
|
extern unsigned int sysctl_tcp_fastopen_blackhole_timeout;
|
2017-04-21 05:45:47 +08:00
|
|
|
void tcp_fastopen_active_disable(struct sock *sk);
|
net/tcp_fastopen: Disable active side TFO in certain scenarios
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: Wei Wang <weiwan@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2017-04-21 05:45:46 +08:00
|
|
|
bool tcp_fastopen_active_should_disable(struct sock *sk);
|
|
|
|
void tcp_fastopen_active_disable_ofo_check(struct sock *sk);
|
|
|
|
void tcp_fastopen_active_timeout_reset(void);
|
|
|
|
|
2016-11-28 15:07:13 +08:00
|
|
|
/* Latencies incurred by various limits for a sender. They are
|
|
|
|
* chronograph-like stats that are mutually exclusive.
|
|
|
|
*/
|
|
|
|
enum tcp_chrono {
|
|
|
|
TCP_CHRONO_UNSPEC,
|
|
|
|
TCP_CHRONO_BUSY, /* Actively sending data (non-empty write queue) */
|
|
|
|
TCP_CHRONO_RWND_LIMITED, /* Stalled by insufficient receive window */
|
|
|
|
TCP_CHRONO_SNDBUF_LIMITED, /* Stalled by insufficient send buffer */
|
|
|
|
__TCP_CHRONO_MAX,
|
|
|
|
};
|
|
|
|
|
|
|
|
void tcp_chrono_start(struct sock *sk, const enum tcp_chrono type);
|
|
|
|
void tcp_chrono_stop(struct sock *sk, const enum tcp_chrono type);
|
|
|
|
|
2007-03-08 04:12:44 +08:00
|
|
|
/* write queue abstraction */
|
|
|
|
static inline void tcp_write_queue_purge(struct sock *sk)
|
|
|
|
{
|
|
|
|
struct sk_buff *skb;
|
|
|
|
|
2016-11-28 15:07:14 +08:00
|
|
|
tcp_chrono_stop(sk, TCP_CHRONO_BUSY);
|
2007-03-08 04:12:44 +08:00
|
|
|
while ((skb = __skb_dequeue(&sk->sk_write_queue)) != NULL)
|
2007-12-31 16:11:19 +08:00
|
|
|
sk_wmem_free_skb(sk, skb);
|
|
|
|
sk_mem_reclaim(sk);
|
2009-12-03 14:24:02 +08:00
|
|
|
tcp_clear_all_retrans_hints(tcp_sk(sk));
|
2007-03-08 04:12:44 +08:00
|
|
|
}
|
|
|
|
|
2011-10-21 17:22:42 +08:00
|
|
|
static inline struct sk_buff *tcp_write_queue_head(const struct sock *sk)
|
2007-03-08 04:12:44 +08:00
|
|
|
{
|
2008-09-23 15:50:13 +08:00
|
|
|
return skb_peek(&sk->sk_write_queue);
|
2007-03-08 04:12:44 +08:00
|
|
|
}
|
|
|
|
|
2011-10-21 17:22:42 +08:00
|
|
|
static inline struct sk_buff *tcp_write_queue_tail(const struct sock *sk)
|
2007-03-08 04:12:44 +08:00
|
|
|
{
|
2008-09-23 15:50:13 +08:00
|
|
|
return skb_peek_tail(&sk->sk_write_queue);
|
2007-03-08 04:12:44 +08:00
|
|
|
}
|
|
|
|
|
2011-10-21 17:22:42 +08:00
|
|
|
static inline struct sk_buff *tcp_write_queue_next(const struct sock *sk,
|
|
|
|
const struct sk_buff *skb)
|
2007-03-08 04:12:44 +08:00
|
|
|
{
|
2008-09-23 15:50:13 +08:00
|
|
|
return skb_queue_next(&sk->sk_write_queue, skb);
|
2007-03-08 04:12:44 +08:00
|
|
|
}
|
|
|
|
|
2011-10-21 17:22:42 +08:00
|
|
|
static inline struct sk_buff *tcp_write_queue_prev(const struct sock *sk,
|
|
|
|
const struct sk_buff *skb)
|
tcp: Try to restore large SKBs while SACK processing
During SACK processing, most of the benefits of TSO are eaten by
the SACK blocks that one-by-one fragment SKBs to MSS sized chunks.
Then we're in problems when cleanup work for them has to be done
when a large cumulative ACK comes. Try to return back to pre-split
state already while more and more SACK info gets discovered by
combining newly discovered SACK areas with the previous skb if
that's SACKed as well.
This approach has a number of benefits:
1) The processing overhead is spread more equally over the RTT
2) Write queue has less skbs to process (affect everything
which has to walk in the queue past the sacked areas)
3) Write queue is consistent whole the time, so no other parts
of TCP has to be aware of this (this was not the case with
some other approach that was, well, quite intrusive all
around).
4) Clean_rtx_queue can release most of the pages using single
put_page instead of previous PAGE_SIZE/mss+1 calls
In case a hole is fully filled by the new SACK block, we attempt
to combine the next skb too which allows construction of skbs
that are even larger than what tso split them to and it handles
hole per on every nth patterns that often occur during slow start
overshoot pretty nicely. Though this to be really useful also
a retransmission would have to get lost since cumulative ACKs
advance one hole at a time in the most typical case.
TODO: handle upwards only merging. That should be rather easy
when segment is fully sacked but I'm leaving that as future
work item (it won't make very large difference anyway since
this current approach already covers quite a lot of normal
cases).
I was earlier thinking of some sophisticated way of tracking
timestamps of the first and the last segment but later on
realized that it won't be that necessary at all to store the
timestamp of the last segment. The cases that can occur are
basically either:
1) ambiguous => no sensible measurement can be taken anyway
2) non-ambiguous is due to reordering => having the timestamp
of the last segment there is just skewing things more off
than does some good since the ack got triggered by one of
the holes (besides some substle issues that would make
determining right hole/skb even harder problem). Anyway,
it has nothing to do with this change then.
I choose to route some abnormal looking cases with goto noop,
some could be handled differently (eg., by stopping the
walking at that skb but again). In general, they either
shouldn't happen at all or are rare enough to make no difference
in practice.
In theory this change (as whole) could cause some macroscale
regression (global) because of cache misses that are taken over
the round-trip time but it gets very likely better because of much
less (local) cache misses per other write queue walkers and the
big recovery clearing cumulative ack.
Worth to note that these benefits would be very easy to get also
without TSO/GSO being on as long as the data is in pages so that
we can merge them. Currently I won't let that happen because
DSACK splitting at fragment that would mess up pcounts due to
sk_can_gso in tcp_set_skb_tso_segs. Once DSACKs fragments gets
avoided, we have some conditions that can be made less strict.
TODO: I will probably have to convert the excessive pointer
passing to struct sacktag_state... :-)
My testing revealed that considerable amount of skbs couldn't
be shifted because they were cloned (most likely still awaiting
tx reclaim)...
[The rest is considering future work instead since I got
repeatably EFAULT to tcpdump's recvfrom when I added
pskb_expand_head to deal with clones, so I separated that
into another, later patch]
...To counter that, I gave up on the fifth advantage:
5) When growing previous SACK block, less allocs for new skbs
are done, basically a new alloc is needed only when new hole
is detected and when the previous skb runs out of frags space
...which now only happens of if reclaim is fast enough to dispose
the clone before the SACK block comes in (the window is RTT long),
otherwise we'll have to alloc some.
With clones being handled I got these numbers (will be somewhat
worse without that), taken with fine-grained mibs:
TCPSackShifted 398
TCPSackMerged 877
TCPSackShiftFallback 320
TCPSACKCOLLAPSEFALLBACKGSO 0
TCPSACKCOLLAPSEFALLBACKSKBBITS 0
TCPSACKCOLLAPSEFALLBACKSKBDATA 0
TCPSACKCOLLAPSEFALLBACKBELOW 0
TCPSACKCOLLAPSEFALLBACKFIRST 1
TCPSACKCOLLAPSEFALLBACKPREVBITS 318
TCPSACKCOLLAPSEFALLBACKMSS 1
TCPSACKCOLLAPSEFALLBACKNOHEAD 0
TCPSACKCOLLAPSEFALLBACKSHIFT 0
TCPSACKCOLLAPSENOOPSEQ 0
TCPSACKCOLLAPSENOOPSMALLPCOUNT 0
TCPSACKCOLLAPSENOOPSMALLLEN 0
TCPSACKCOLLAPSEHOLE 12
Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
2008-11-25 13:20:15 +08:00
|
|
|
{
|
|
|
|
return skb_queue_prev(&sk->sk_write_queue, skb);
|
|
|
|
}
|
|
|
|
|
2007-03-08 04:12:44 +08:00
|
|
|
#define tcp_for_write_queue(skb, sk) \
|
2008-09-23 15:50:13 +08:00
|
|
|
skb_queue_walk(&(sk)->sk_write_queue, skb)
|
2007-03-08 04:12:44 +08:00
|
|
|
|
|
|
|
#define tcp_for_write_queue_from(skb, sk) \
|
2008-09-23 15:50:13 +08:00
|
|
|
skb_queue_walk_from(&(sk)->sk_write_queue, skb)
|
2007-03-08 04:12:44 +08:00
|
|
|
|
2007-12-02 06:48:02 +08:00
|
|
|
#define tcp_for_write_queue_from_safe(skb, tmp, sk) \
|
2008-09-23 15:50:13 +08:00
|
|
|
skb_queue_walk_from_safe(&(sk)->sk_write_queue, skb, tmp)
|
2007-12-02 06:48:02 +08:00
|
|
|
|
2011-10-21 17:22:42 +08:00
|
|
|
static inline struct sk_buff *tcp_send_head(const struct sock *sk)
|
2007-03-08 04:12:44 +08:00
|
|
|
{
|
|
|
|
return sk->sk_send_head;
|
|
|
|
}
|
|
|
|
|
2008-09-23 15:50:13 +08:00
|
|
|
static inline bool tcp_skb_is_last(const struct sock *sk,
|
|
|
|
const struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
return skb_queue_is_last(&sk->sk_write_queue, skb);
|
|
|
|
}
|
|
|
|
|
2011-10-21 17:22:42 +08:00
|
|
|
static inline void tcp_advance_send_head(struct sock *sk, const struct sk_buff *skb)
|
2007-03-08 04:12:44 +08:00
|
|
|
{
|
2008-09-23 15:50:13 +08:00
|
|
|
if (tcp_skb_is_last(sk, skb))
|
2007-03-08 04:12:44 +08:00
|
|
|
sk->sk_send_head = NULL;
|
2008-09-23 15:50:13 +08:00
|
|
|
else
|
|
|
|
sk->sk_send_head = tcp_write_queue_next(sk, skb);
|
2007-03-08 04:12:44 +08:00
|
|
|
}
|
|
|
|
|
|
|
|
static inline void tcp_check_send_head(struct sock *sk, struct sk_buff *skb_unlinked)
|
|
|
|
{
|
2016-11-28 15:07:14 +08:00
|
|
|
if (sk->sk_send_head == skb_unlinked) {
|
2007-03-08 04:12:44 +08:00
|
|
|
sk->sk_send_head = NULL;
|
2016-11-28 15:07:14 +08:00
|
|
|
tcp_chrono_stop(sk, TCP_CHRONO_BUSY);
|
|
|
|
}
|
2016-08-17 20:56:26 +08:00
|
|
|
if (tcp_sk(sk)->highest_sack == skb_unlinked)
|
|
|
|
tcp_sk(sk)->highest_sack = NULL;
|
2007-03-08 04:12:44 +08:00
|
|
|
}
|
|
|
|
|
|
|
|
static inline void tcp_init_send_head(struct sock *sk)
|
|
|
|
{
|
|
|
|
sk->sk_send_head = NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
static inline void __tcp_add_write_queue_tail(struct sock *sk, struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
__skb_queue_tail(&sk->sk_write_queue, skb);
|
|
|
|
}
|
|
|
|
|
|
|
|
static inline void tcp_add_write_queue_tail(struct sock *sk, struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
__tcp_add_write_queue_tail(sk, skb);
|
|
|
|
|
|
|
|
/* Queue it, remembering where we must start sending. */
|
2007-12-02 06:48:06 +08:00
|
|
|
if (sk->sk_send_head == NULL) {
|
2007-03-08 04:12:44 +08:00
|
|
|
sk->sk_send_head = skb;
|
2016-11-28 15:07:14 +08:00
|
|
|
tcp_chrono_start(sk, TCP_CHRONO_BUSY);
|
2007-12-02 06:48:06 +08:00
|
|
|
|
|
|
|
if (tcp_sk(sk)->highest_sack == NULL)
|
|
|
|
tcp_sk(sk)->highest_sack = skb;
|
|
|
|
}
|
2007-03-08 04:12:44 +08:00
|
|
|
}
|
|
|
|
|
|
|
|
static inline void __tcp_add_write_queue_head(struct sock *sk, struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
__skb_queue_head(&sk->sk_write_queue, skb);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Insert buff after skb on the write queue of sk. */
|
|
|
|
static inline void tcp_insert_write_queue_after(struct sk_buff *skb,
|
|
|
|
struct sk_buff *buff,
|
|
|
|
struct sock *sk)
|
|
|
|
{
|
2008-04-14 15:05:09 +08:00
|
|
|
__skb_queue_after(&sk->sk_write_queue, skb, buff);
|
2007-03-08 04:12:44 +08:00
|
|
|
}
|
|
|
|
|
2008-09-22 12:28:51 +08:00
|
|
|
/* Insert new before skb on the write queue of sk. */
|
2007-03-08 04:12:44 +08:00
|
|
|
static inline void tcp_insert_write_queue_before(struct sk_buff *new,
|
|
|
|
struct sk_buff *skb,
|
|
|
|
struct sock *sk)
|
|
|
|
{
|
2008-09-22 12:28:51 +08:00
|
|
|
__skb_queue_before(&sk->sk_write_queue, skb, new);
|
2007-11-20 15:24:09 +08:00
|
|
|
|
|
|
|
if (sk->sk_send_head == skb)
|
|
|
|
sk->sk_send_head = new;
|
2007-03-08 04:12:44 +08:00
|
|
|
}
|
|
|
|
|
|
|
|
static inline void tcp_unlink_write_queue(struct sk_buff *skb, struct sock *sk)
|
|
|
|
{
|
|
|
|
__skb_unlink(skb, &sk->sk_write_queue);
|
|
|
|
}
|
|
|
|
|
2012-05-17 07:15:34 +08:00
|
|
|
static inline bool tcp_write_queue_empty(struct sock *sk)
|
2007-03-08 04:12:44 +08:00
|
|
|
{
|
|
|
|
return skb_queue_empty(&sk->sk_write_queue);
|
|
|
|
}
|
|
|
|
|
2009-12-09 06:26:13 +08:00
|
|
|
static inline void tcp_push_pending_frames(struct sock *sk)
|
|
|
|
{
|
|
|
|
if (tcp_send_head(sk)) {
|
|
|
|
struct tcp_sock *tp = tcp_sk(sk);
|
|
|
|
|
|
|
|
__tcp_push_pending_frames(sk, tcp_current_mss(sk), tp->nonagle);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2012-02-28 06:52:52 +08:00
|
|
|
/* Start sequence of the skb just after the highest skb with SACKed
|
|
|
|
* bit, valid only if sacked_out > 0 or when the caller has ensured
|
|
|
|
* validity by itself.
|
2007-11-16 11:41:46 +08:00
|
|
|
*/
|
|
|
|
static inline u32 tcp_highest_sack_seq(struct tcp_sock *tp)
|
|
|
|
{
|
|
|
|
if (!tp->sacked_out)
|
|
|
|
return tp->snd_una;
|
2007-12-02 06:48:06 +08:00
|
|
|
|
|
|
|
if (tp->highest_sack == NULL)
|
|
|
|
return tp->snd_nxt;
|
|
|
|
|
2007-11-16 11:41:46 +08:00
|
|
|
return TCP_SKB_CB(tp->highest_sack)->seq;
|
|
|
|
}
|
|
|
|
|
2007-12-02 06:48:06 +08:00
|
|
|
static inline void tcp_advance_highest_sack(struct sock *sk, struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
tcp_sk(sk)->highest_sack = tcp_skb_is_last(sk, skb) ? NULL :
|
|
|
|
tcp_write_queue_next(sk, skb);
|
|
|
|
}
|
|
|
|
|
|
|
|
static inline struct sk_buff *tcp_highest_sack(struct sock *sk)
|
|
|
|
{
|
|
|
|
return tcp_sk(sk)->highest_sack;
|
|
|
|
}
|
|
|
|
|
|
|
|
static inline void tcp_highest_sack_reset(struct sock *sk)
|
|
|
|
{
|
|
|
|
tcp_sk(sk)->highest_sack = tcp_write_queue_head(sk);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Called when old skb is about to be deleted (to be combined with new skb) */
|
|
|
|
static inline void tcp_highest_sack_combine(struct sock *sk,
|
|
|
|
struct sk_buff *old,
|
|
|
|
struct sk_buff *new)
|
|
|
|
{
|
|
|
|
if (tcp_sk(sk)->sacked_out && (old == tcp_sk(sk)->highest_sack))
|
|
|
|
tcp_sk(sk)->highest_sack = new;
|
|
|
|
}
|
|
|
|
|
2015-12-22 04:29:24 +08:00
|
|
|
/* This helper checks if socket has IP_TRANSPARENT set */
|
|
|
|
static inline bool inet_sk_transparent(const struct sock *sk)
|
|
|
|
{
|
|
|
|
switch (sk->sk_state) {
|
|
|
|
case TCP_TIME_WAIT:
|
|
|
|
return inet_twsk(sk)->tw_transparent;
|
|
|
|
case TCP_NEW_SYN_RECV:
|
|
|
|
return inet_rsk(inet_reqsk(sk))->no_srccheck;
|
|
|
|
}
|
|
|
|
return inet_sk(sk)->transparent;
|
|
|
|
}
|
|
|
|
|
2010-02-18 10:45:45 +08:00
|
|
|
/* Determines whether this is a thin stream (which may suffer from
|
|
|
|
* increased latency). Used to trigger latency-reducing mechanisms.
|
|
|
|
*/
|
2012-05-17 07:15:34 +08:00
|
|
|
static inline bool tcp_stream_is_thin(struct tcp_sock *tp)
|
2010-02-18 10:45:45 +08:00
|
|
|
{
|
|
|
|
return tp->packets_out < 4 && !tcp_in_initial_slowstart(tp);
|
|
|
|
}
|
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
/* /proc */
|
|
|
|
enum tcp_seq_states {
|
|
|
|
TCP_SEQ_STATE_LISTENING,
|
|
|
|
TCP_SEQ_STATE_ESTABLISHED,
|
|
|
|
};
|
|
|
|
|
2011-10-30 14:46:30 +08:00
|
|
|
int tcp_seq_open(struct inode *inode, struct file *file);
|
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
struct tcp_seq_afinfo {
|
2011-10-30 14:46:30 +08:00
|
|
|
char *name;
|
|
|
|
sa_family_t family;
|
|
|
|
const struct file_operations *seq_fops;
|
|
|
|
struct seq_operations seq_ops;
|
2005-04-17 06:20:36 +08:00
|
|
|
};
|
|
|
|
|
|
|
|
struct tcp_iter_state {
|
2008-04-14 13:11:14 +08:00
|
|
|
struct seq_net_private p;
|
2005-04-17 06:20:36 +08:00
|
|
|
sa_family_t family;
|
|
|
|
enum tcp_seq_states state;
|
|
|
|
struct sock *syn_wait_sk;
|
2012-05-24 15:10:10 +08:00
|
|
|
int bucket, offset, sbucket, num;
|
2010-06-07 15:43:42 +08:00
|
|
|
loff_t last_pos;
|
2005-04-17 06:20:36 +08:00
|
|
|
};
|
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp_proc_register(struct net *net, struct tcp_seq_afinfo *afinfo);
|
|
|
|
void tcp_proc_unregister(struct net *net, struct tcp_seq_afinfo *afinfo);
|
2005-04-17 06:20:36 +08:00
|
|
|
|
2005-08-16 13:18:02 +08:00
|
|
|
extern struct request_sock_ops tcp_request_sock_ops;
|
2008-02-08 13:49:26 +08:00
|
|
|
extern struct request_sock_ops tcp6_request_sock_ops;
|
2005-08-16 13:18:02 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_v4_destroy_sock(struct sock *sk);
|
2005-08-16 13:18:02 +08:00
|
|
|
|
2013-10-19 01:36:17 +08:00
|
|
|
struct sk_buff *tcp_gso_segment(struct sk_buff *skb,
|
2013-09-24 02:33:32 +08:00
|
|
|
netdev_features_t features);
|
|
|
|
struct sk_buff **tcp_gro_receive(struct sk_buff **head, struct sk_buff *skb);
|
|
|
|
int tcp_gro_complete(struct sk_buff *skb);
|
2013-06-07 13:11:46 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
void __tcp_v4_send_check(struct sk_buff *skb, __be32 saddr, __be32 daddr);
|
2006-06-22 18:02:40 +08:00
|
|
|
|
tcp: TCP_NOTSENT_LOWAT socket option
Idea of this patch is to add optional limitation of number of
unsent bytes in TCP sockets, to reduce usage of kernel memory.
TCP receiver might announce a big window, and TCP sender autotuning
might allow a large amount of bytes in write queue, but this has little
performance impact if a large part of this buffering is wasted :
Write queue needs to be large only to deal with large BDP, not
necessarily to cope with scheduling delays (incoming ACKS make room
for the application to queue more bytes)
For most workloads, using a value of 128 KB or less is OK to give
applications enough time to react to POLLOUT events in time
(or being awaken in a blocking sendmsg())
This patch adds two ways to set the limit :
1) Per socket option TCP_NOTSENT_LOWAT
2) A sysctl (/proc/sys/net/ipv4/tcp_notsent_lowat) for sockets
not using TCP_NOTSENT_LOWAT socket option (or setting a zero value)
Default value being UINT_MAX (0xFFFFFFFF), meaning this has no effect.
This changes poll()/select()/epoll() to report POLLOUT
only if number of unsent bytes is below tp->nosent_lowat
Note this might increase number of sendmsg()/sendfile() calls
when using non blocking sockets,
and increase number of context switches for blocking sockets.
Note this is not related to SO_SNDLOWAT (as SO_SNDLOWAT is
defined as :
Specify the minimum number of bytes in the buffer until
the socket layer will pass the data to the protocol)
Tested:
netperf sessions, and watching /proc/net/protocols "memory" column for TCP
With 200 concurrent netperf -t TCP_STREAM sessions, amount of kernel memory
used by TCP buffers shrinks by ~55 % (20567 pages instead of 45458)
lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols
TCPv6 1880 2 45458 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y
TCP 1696 508 45458 no 208 yes kernel y y y y y y y y y y y y y n y y y y y
lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols
TCPv6 1880 2 20567 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y
TCP 1696 508 20567 no 208 yes kernel y y y y y y y y y y y y y n y y y y y
Using 128KB has no bad effect on the throughput or cpu usage
of a single flow, although there is an increase of context switches.
A bonus is that we hold socket lock for a shorter amount
of time and should improve latencies of ACK processing.
lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3
OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf.
Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service
Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand
Size Size Size (sec) Util Util Util Util Demand Demand Units
Final Final % Method % Method
1651584 6291456 16384 20.00 17447.90 10^6bits/s 3.13 S -1.00 U 0.353 -1.000 usec/KB
Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3':
412,514 context-switches
200.034645535 seconds time elapsed
lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3
OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf.
Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service
Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand
Size Size Size (sec) Util Util Util Util Demand Demand Units
Final Final % Method % Method
1593240 6291456 16384 20.00 17321.16 10^6bits/s 3.35 S -1.00 U 0.381 -1.000 usec/KB
Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3':
2,675,818 context-switches
200.029651391 seconds time elapsed
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-By: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2013-07-23 11:27:07 +08:00
|
|
|
static inline u32 tcp_notsent_lowat(const struct tcp_sock *tp)
|
|
|
|
{
|
2016-02-03 15:46:57 +08:00
|
|
|
struct net *net = sock_net((struct sock *)tp);
|
|
|
|
return tp->notsent_lowat ?: net->ipv4.sysctl_tcp_notsent_lowat;
|
tcp: TCP_NOTSENT_LOWAT socket option
Idea of this patch is to add optional limitation of number of
unsent bytes in TCP sockets, to reduce usage of kernel memory.
TCP receiver might announce a big window, and TCP sender autotuning
might allow a large amount of bytes in write queue, but this has little
performance impact if a large part of this buffering is wasted :
Write queue needs to be large only to deal with large BDP, not
necessarily to cope with scheduling delays (incoming ACKS make room
for the application to queue more bytes)
For most workloads, using a value of 128 KB or less is OK to give
applications enough time to react to POLLOUT events in time
(or being awaken in a blocking sendmsg())
This patch adds two ways to set the limit :
1) Per socket option TCP_NOTSENT_LOWAT
2) A sysctl (/proc/sys/net/ipv4/tcp_notsent_lowat) for sockets
not using TCP_NOTSENT_LOWAT socket option (or setting a zero value)
Default value being UINT_MAX (0xFFFFFFFF), meaning this has no effect.
This changes poll()/select()/epoll() to report POLLOUT
only if number of unsent bytes is below tp->nosent_lowat
Note this might increase number of sendmsg()/sendfile() calls
when using non blocking sockets,
and increase number of context switches for blocking sockets.
Note this is not related to SO_SNDLOWAT (as SO_SNDLOWAT is
defined as :
Specify the minimum number of bytes in the buffer until
the socket layer will pass the data to the protocol)
Tested:
netperf sessions, and watching /proc/net/protocols "memory" column for TCP
With 200 concurrent netperf -t TCP_STREAM sessions, amount of kernel memory
used by TCP buffers shrinks by ~55 % (20567 pages instead of 45458)
lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols
TCPv6 1880 2 45458 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y
TCP 1696 508 45458 no 208 yes kernel y y y y y y y y y y y y y n y y y y y
lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols
TCPv6 1880 2 20567 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y
TCP 1696 508 20567 no 208 yes kernel y y y y y y y y y y y y y n y y y y y
Using 128KB has no bad effect on the throughput or cpu usage
of a single flow, although there is an increase of context switches.
A bonus is that we hold socket lock for a shorter amount
of time and should improve latencies of ACK processing.
lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3
OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf.
Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service
Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand
Size Size Size (sec) Util Util Util Util Demand Demand Units
Final Final % Method % Method
1651584 6291456 16384 20.00 17447.90 10^6bits/s 3.13 S -1.00 U 0.353 -1.000 usec/KB
Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3':
412,514 context-switches
200.034645535 seconds time elapsed
lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3
OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf.
Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service
Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand
Size Size Size (sec) Util Util Util Util Demand Demand Units
Final Final % Method % Method
1593240 6291456 16384 20.00 17321.16 10^6bits/s 3.35 S -1.00 U 0.381 -1.000 usec/KB
Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3':
2,675,818 context-switches
200.029651391 seconds time elapsed
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-By: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2013-07-23 11:27:07 +08:00
|
|
|
}
|
|
|
|
|
|
|
|
static inline bool tcp_stream_memory_free(const struct sock *sk)
|
|
|
|
{
|
|
|
|
const struct tcp_sock *tp = tcp_sk(sk);
|
|
|
|
u32 notsent_bytes = tp->write_seq - tp->snd_nxt;
|
|
|
|
|
|
|
|
return notsent_bytes < tcp_notsent_lowat(tp);
|
|
|
|
}
|
|
|
|
|
2005-08-16 13:18:02 +08:00
|
|
|
#ifdef CONFIG_PROC_FS
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcp4_proc_init(void);
|
|
|
|
void tcp4_proc_exit(void);
|
2005-08-16 13:18:02 +08:00
|
|
|
#endif
|
|
|
|
|
2015-09-25 22:39:23 +08:00
|
|
|
int tcp_rtx_synack(const struct sock *sk, struct request_sock *req);
|
2014-06-25 22:10:02 +08:00
|
|
|
int tcp_conn_request(struct request_sock_ops *rsk_ops,
|
|
|
|
const struct tcp_request_sock_ops *af_ops,
|
|
|
|
struct sock *sk, struct sk_buff *skb);
|
2014-06-25 22:09:59 +08:00
|
|
|
|
2006-11-15 11:07:45 +08:00
|
|
|
/* TCP af-specific functions */
|
|
|
|
struct tcp_sock_af_ops {
|
|
|
|
#ifdef CONFIG_TCP_MD5SIG
|
2015-09-25 22:39:15 +08:00
|
|
|
struct tcp_md5sig_key *(*md5_lookup) (const struct sock *sk,
|
2015-03-25 06:58:56 +08:00
|
|
|
const struct sock *addr_sk);
|
2015-03-25 06:58:55 +08:00
|
|
|
int (*calc_md5_hash)(char *location,
|
|
|
|
const struct tcp_md5sig_key *md5,
|
|
|
|
const struct sock *sk,
|
|
|
|
const struct sk_buff *skb);
|
|
|
|
int (*md5_parse)(struct sock *sk,
|
|
|
|
char __user *optval,
|
|
|
|
int optlen);
|
2006-11-15 11:07:45 +08:00
|
|
|
#endif
|
|
|
|
};
|
|
|
|
|
|
|
|
struct tcp_request_sock_ops {
|
2014-06-25 22:10:00 +08:00
|
|
|
u16 mss_clamp;
|
2006-11-15 11:07:45 +08:00
|
|
|
#ifdef CONFIG_TCP_MD5SIG
|
2015-09-25 22:39:15 +08:00
|
|
|
struct tcp_md5sig_key *(*req_md5_lookup)(const struct sock *sk,
|
2015-03-25 06:58:56 +08:00
|
|
|
const struct sock *addr_sk);
|
2015-03-25 06:58:55 +08:00
|
|
|
int (*calc_md5_hash) (char *location,
|
|
|
|
const struct tcp_md5sig_key *md5,
|
|
|
|
const struct sock *sk,
|
|
|
|
const struct sk_buff *skb);
|
2006-11-15 11:07:45 +08:00
|
|
|
#endif
|
2015-09-25 22:39:08 +08:00
|
|
|
void (*init_req)(struct request_sock *req,
|
|
|
|
const struct sock *sk_listener,
|
2014-06-25 22:09:53 +08:00
|
|
|
struct sk_buff *skb);
|
2014-06-25 22:09:54 +08:00
|
|
|
#ifdef CONFIG_SYN_COOKIES
|
2015-09-29 22:42:49 +08:00
|
|
|
__u32 (*cookie_init_seq)(const struct sk_buff *skb,
|
2014-06-25 22:09:54 +08:00
|
|
|
__u16 *mss);
|
|
|
|
#endif
|
2015-09-29 22:42:50 +08:00
|
|
|
struct dst_entry *(*route_req)(const struct sock *sk, struct flowi *fl,
|
2017-03-16 04:30:46 +08:00
|
|
|
const struct request_sock *req);
|
2017-05-05 21:56:54 +08:00
|
|
|
u32 (*init_seq)(const struct sk_buff *skb);
|
|
|
|
u32 (*init_ts_off)(const struct sk_buff *skb);
|
2015-09-25 22:39:21 +08:00
|
|
|
int (*send_synack)(const struct sock *sk, struct dst_entry *dst,
|
2014-06-25 22:09:58 +08:00
|
|
|
struct flowi *fl, struct request_sock *req,
|
2015-10-17 04:00:01 +08:00
|
|
|
struct tcp_fastopen_cookie *foc,
|
2016-04-14 13:05:39 +08:00
|
|
|
enum tcp_synack_type synack_type);
|
2006-11-15 11:07:45 +08:00
|
|
|
};
|
|
|
|
|
2014-06-25 22:09:54 +08:00
|
|
|
#ifdef CONFIG_SYN_COOKIES
|
|
|
|
static inline __u32 cookie_init_sequence(const struct tcp_request_sock_ops *ops,
|
2015-09-29 22:42:49 +08:00
|
|
|
const struct sock *sk, struct sk_buff *skb,
|
2014-06-25 22:09:54 +08:00
|
|
|
__u16 *mss)
|
|
|
|
{
|
2015-09-29 22:42:49 +08:00
|
|
|
tcp_synq_overflow(sk);
|
2016-04-28 07:44:39 +08:00
|
|
|
__NET_INC_STATS(sock_net(sk), LINUX_MIB_SYNCOOKIESSENT);
|
2015-09-29 22:42:49 +08:00
|
|
|
return ops->cookie_init_seq(skb, mss);
|
2014-06-25 22:09:54 +08:00
|
|
|
}
|
|
|
|
#else
|
|
|
|
static inline __u32 cookie_init_sequence(const struct tcp_request_sock_ops *ops,
|
2015-09-29 22:42:49 +08:00
|
|
|
const struct sock *sk, struct sk_buff *skb,
|
2014-06-25 22:09:54 +08:00
|
|
|
__u16 *mss)
|
|
|
|
{
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
#endif
|
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
int tcpv4_offload_init(void);
|
2013-06-07 13:11:46 +08:00
|
|
|
|
2013-09-24 02:33:32 +08:00
|
|
|
void tcp_v4_init(void);
|
|
|
|
void tcp_init(void);
|
2005-08-16 13:18:02 +08:00
|
|
|
|
2015-10-17 12:57:46 +08:00
|
|
|
/* tcp_recovery.c */
|
2017-04-26 01:15:34 +08:00
|
|
|
extern void tcp_rack_mark_lost(struct sock *sk);
|
2017-01-13 14:11:34 +08:00
|
|
|
extern void tcp_rack_advance(struct tcp_sock *tp, u8 sacked, u32 end_seq,
|
2017-05-17 05:00:14 +08:00
|
|
|
u64 xmit_time);
|
2017-01-13 14:11:33 +08:00
|
|
|
extern void tcp_rack_reo_timeout(struct sock *sk);
|
2015-10-17 12:57:46 +08:00
|
|
|
|
2014-10-16 05:33:21 +08:00
|
|
|
/*
|
|
|
|
* Save and compile IPv4 options, return a pointer to it
|
|
|
|
*/
|
|
|
|
static inline struct ip_options_rcu *tcp_v4_save_options(struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
const struct ip_options *opt = &TCP_SKB_CB(skb)->header.h4.opt;
|
|
|
|
struct ip_options_rcu *dopt = NULL;
|
|
|
|
|
2014-10-16 05:33:22 +08:00
|
|
|
if (opt->optlen) {
|
2014-10-16 05:33:21 +08:00
|
|
|
int opt_size = sizeof(*dopt) + opt->optlen;
|
|
|
|
|
|
|
|
dopt = kmalloc(opt_size, GFP_ATOMIC);
|
|
|
|
if (dopt && __ip_options_echo(&dopt->opt, skb, opt)) {
|
|
|
|
kfree(dopt);
|
|
|
|
dopt = NULL;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
return dopt;
|
|
|
|
}
|
|
|
|
|
tcp: do not pace pure ack packets
When we added pacing to TCP, we decided to let sch_fq take care
of actual pacing.
All TCP had to do was to compute sk->pacing_rate using simple formula:
sk->pacing_rate = 2 * cwnd * mss / rtt
It works well for senders (bulk flows), but not very well for receivers
or even RPC :
cwnd on the receiver can be less than 10, rtt can be around 100ms, so we
can end up pacing ACK packets, slowing down the sender.
Really, only the sender should pace, according to its own logic.
Instead of adding a new bit in skb, or call yet another flow
dissection, we tweak skb->truesize to a small value (2), and
we instruct sch_fq to use new helper and not pace pure ack.
Note this also helps TCP small queue, as ack packets present
in qdisc/NIC do not prevent sending a data packet (RPC workload)
This helps to reduce tx completion overhead, ack packets can use regular
sock_wfree() instead of tcp_wfree() which is a bit more expensive.
This has no impact in the case packets are sent to loopback interface,
as we do not coalesce ack packets (were we would detect skb->truesize
lie)
In case netem (with a delay) is used, skb_orphan_partial() also sets
skb->truesize to 1.
This patch is a combination of two patches we used for about one year at
Google.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-02-04 10:31:53 +08:00
|
|
|
/* locally generated TCP pure ACKs have skb->truesize == 2
|
|
|
|
* (check tcp_send_ack() in net/ipv4/tcp_output.c )
|
|
|
|
* This is much faster than dissecting the packet to find out.
|
|
|
|
* (Think of GRE encapsulations, IPv4, IPv6, ...)
|
|
|
|
*/
|
|
|
|
static inline bool skb_is_tcp_pure_ack(const struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
return skb->truesize == 2;
|
|
|
|
}
|
|
|
|
|
|
|
|
static inline void skb_set_tcp_pure_ack(struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
skb->truesize = 2;
|
|
|
|
}
|
|
|
|
|
2016-03-08 06:11:05 +08:00
|
|
|
static inline int tcp_inq(struct sock *sk)
|
|
|
|
{
|
|
|
|
struct tcp_sock *tp = tcp_sk(sk);
|
|
|
|
int answ;
|
|
|
|
|
|
|
|
if ((1 << sk->sk_state) & (TCPF_SYN_SENT | TCPF_SYN_RECV)) {
|
|
|
|
answ = 0;
|
|
|
|
} else if (sock_flag(sk, SOCK_URGINLINE) ||
|
|
|
|
!tp->urg_data ||
|
|
|
|
before(tp->urg_seq, tp->copied_seq) ||
|
|
|
|
!before(tp->urg_seq, tp->rcv_nxt)) {
|
|
|
|
|
|
|
|
answ = tp->rcv_nxt - tp->copied_seq;
|
|
|
|
|
|
|
|
/* Subtract 1, if FIN was received */
|
|
|
|
if (answ && sock_flag(sk, SOCK_DONE))
|
|
|
|
answ--;
|
|
|
|
} else {
|
|
|
|
answ = tp->urg_seq - tp->copied_seq;
|
|
|
|
}
|
|
|
|
|
|
|
|
return answ;
|
|
|
|
}
|
|
|
|
|
2016-08-29 05:43:18 +08:00
|
|
|
int tcp_peek_len(struct socket *sock);
|
|
|
|
|
2016-03-15 01:52:15 +08:00
|
|
|
static inline void tcp_segs_in(struct tcp_sock *tp, const struct sk_buff *skb)
|
|
|
|
{
|
|
|
|
u16 segs_in;
|
|
|
|
|
|
|
|
segs_in = max_t(u16, 1, skb_shinfo(skb)->gso_segs);
|
|
|
|
tp->segs_in += segs_in;
|
|
|
|
if (skb->len > tcp_hdrlen(skb))
|
|
|
|
tp->data_segs_in += segs_in;
|
|
|
|
}
|
|
|
|
|
2016-04-01 23:52:20 +08:00
|
|
|
/*
|
|
|
|
* TCP listen path runs lockless.
|
|
|
|
* We forced "struct sock" to be const qualified to make sure
|
|
|
|
* we don't modify one of its field by mistake.
|
|
|
|
* Here, we increment sk_drops which is an atomic_t, so we can safely
|
|
|
|
* make sock writable again.
|
|
|
|
*/
|
|
|
|
static inline void tcp_listendrop(const struct sock *sk)
|
|
|
|
{
|
|
|
|
atomic_inc(&((struct sock *)sk)->sk_drops);
|
2016-04-28 07:44:39 +08:00
|
|
|
__NET_INC_STATS(sock_net(sk), LINUX_MIB_LISTENDROPS);
|
2016-04-01 23:52:20 +08:00
|
|
|
}
|
|
|
|
|
2017-05-16 19:24:36 +08:00
|
|
|
enum hrtimer_restart tcp_pace_kick(struct hrtimer *timer);
|
|
|
|
|
2005-04-17 06:20:36 +08:00
|
|
|
#endif /* _TCP_H */
|