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tcp_input.c
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/*
* INET An implementation of the TCP/IP protocol suite for the LINUX
* operating system. INET is implemented using the BSD Socket
* interface as the means of communication with the user level.
*
* Implementation of the Transmission Control Protocol(TCP).
*
* Authors: Ross Biro
* Fred N. van Kempen, <waltje@uWalt.NL.Mugnet.ORG>
* Mark Evans, <evansmp@uhura.aston.ac.uk>
* Corey Minyard <wf-rch!minyard@relay.EU.net>
* Florian La Roche, <flla@stud.uni-sb.de>
* Charles Hedrick, <hedrick@klinzhai.rutgers.edu>
* Linus Torvalds, <torvalds@cs.helsinki.fi>
* Alan Cox, <gw4pts@gw4pts.ampr.org>
* Matthew Dillon, <dillon@apollo.west.oic.com>
* Arnt Gulbrandsen, <agulbra@nvg.unit.no>
* Jorge Cwik, <jorge@laser.satlink.net>
*/
/*
* Changes:
* Pedro Roque : Fast Retransmit/Recovery.
* Two receive queues.
* Retransmit queue handled by TCP.
* Better retransmit timer handling.
* New congestion avoidance.
* Header prediction.
* Variable renaming.
*
* Eric : Fast Retransmit.
* Randy Scott : MSS option defines.
* Eric Schenk : Fixes to slow start algorithm.
* Eric Schenk : Yet another double ACK bug.
* Eric Schenk : Delayed ACK bug fixes.
* Eric Schenk : Floyd style fast retrans war avoidance.
* David S. Miller : Don't allow zero congestion window.
* Eric Schenk : Fix retransmitter so that it sends
* next packet on ack of previous packet.
* Andi Kleen : Moved open_request checking here
* and process RSTs for open_requests.
* Andi Kleen : Better prune_queue, and other fixes.
* Andrey Savochkin: Fix RTT measurements in the presence of
* timestamps.
* Andrey Savochkin: Check sequence numbers correctly when
* removing SACKs due to in sequence incoming
* data segments.
* Andi Kleen: Make sure we never ack data there is not
* enough room for. Also make this condition
* a fatal error if it might still happen.
* Andi Kleen: Add tcp_measure_rcv_mss to make
* connections with MSS<min(MTU,ann. MSS)
* work without delayed acks.
* Andi Kleen: Process packets with PSH set in the
* fast path.
* J Hadi Salim: ECN support
* Andrei Gurtov,
* Pasi Sarolahti,
* Panu Kuhlberg: Experimental audit of TCP (re)transmission
* engine. Lots of bugs are found.
* Pasi Sarolahti: F-RTO for dealing with spurious RTOs
*/
#define pr_fmt(fmt) "TCP: " fmt
#include <linux/mm.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/sysctl.h>
#include <linux/kernel.h>
#include <net/dst.h>
#include <net/tcp.h>
#include <net/inet_common.h>
#include <linux/ipsec.h>
#include <asm/unaligned.h>
#include <net/netdma.h>
int sysctl_tcp_timestamps __read_mostly = 1;
int sysctl_tcp_window_scaling __read_mostly = 1;
int sysctl_tcp_sack __read_mostly = 1;
int sysctl_tcp_fack __read_mostly = 1;
int sysctl_tcp_reordering __read_mostly = TCP_FASTRETRANS_THRESH;
EXPORT_SYMBOL(sysctl_tcp_reordering);
int sysctl_tcp_dsack __read_mostly = 1;
int sysctl_tcp_app_win __read_mostly = 31;
int sysctl_tcp_adv_win_scale __read_mostly = 1;
EXPORT_SYMBOL(sysctl_tcp_adv_win_scale);
/* rfc5961 challenge ack rate limiting */
int sysctl_tcp_challenge_ack_limit = 100;
int sysctl_tcp_stdurg __read_mostly;
int sysctl_tcp_rfc1337 __read_mostly;
int sysctl_tcp_max_orphans __read_mostly = NR_FILE;
int sysctl_tcp_frto __read_mostly = 2;
int sysctl_tcp_thin_dupack __read_mostly;
int sysctl_tcp_moderate_rcvbuf __read_mostly = 1;
int sysctl_tcp_early_retrans __read_mostly = 3;
#define FLAG_DATA 0x01 /* Incoming frame contained data. */
#define FLAG_WIN_UPDATE 0x02 /* Incoming ACK was a window update. */
#define FLAG_DATA_ACKED 0x04 /* This ACK acknowledged new data. */
#define FLAG_RETRANS_DATA_ACKED 0x08 /* "" "" some of which was retransmitted. */
#define FLAG_SYN_ACKED 0x10 /* This ACK acknowledged SYN. */
#define FLAG_DATA_SACKED 0x20 /* New SACK. */
#define FLAG_ECE 0x40 /* ECE in this ACK */
#define FLAG_SLOWPATH 0x100 /* Do not skip RFC checks for window update.*/
#define FLAG_ORIG_SACK_ACKED 0x200 /* Never retransmitted data are (s)acked */
#define FLAG_SND_UNA_ADVANCED 0x400 /* Snd_una was changed (!= FLAG_DATA_ACKED) */
#define FLAG_DSACKING_ACK 0x800 /* SACK blocks contained D-SACK info */
#define FLAG_SACK_RENEGING 0x2000 /* snd_una advanced to a sacked seq */
#define FLAG_UPDATE_TS_RECENT 0x4000 /* tcp_replace_ts_recent() */
#define FLAG_ACKED (FLAG_DATA_ACKED|FLAG_SYN_ACKED)
#define FLAG_NOT_DUP (FLAG_DATA|FLAG_WIN_UPDATE|FLAG_ACKED)
#define FLAG_CA_ALERT (FLAG_DATA_SACKED|FLAG_ECE)
#define FLAG_FORWARD_PROGRESS (FLAG_ACKED|FLAG_DATA_SACKED)
#define TCP_REMNANT (TCP_FLAG_FIN|TCP_FLAG_URG|TCP_FLAG_SYN|TCP_FLAG_PSH)
#define TCP_HP_BITS (~(TCP_RESERVED_BITS|TCP_FLAG_PSH))
/* Adapt the MSS value used to make delayed ack decision to the
* real world.
*/
static void tcp_measure_rcv_mss(struct sock *sk, const struct sk_buff *skb)
{
struct inet_connection_sock *icsk = inet_csk(sk);
const unsigned int lss = icsk->icsk_ack.last_seg_size;
unsigned int len;
icsk->icsk_ack.last_seg_size = 0;
/* skb->len may jitter because of SACKs, even if peer
* sends good full-sized frames.
*/
len = skb_shinfo(skb)->gso_size ? : skb->len;
if (len >= icsk->icsk_ack.rcv_mss) {
icsk->icsk_ack.rcv_mss = len;
} else {
/* Otherwise, we make more careful check taking into account,
* that SACKs block is variable.
*
* "len" is invariant segment length, including TCP header.
*/
len += skb->data - skb_transport_header(skb);
if (len >= TCP_MSS_DEFAULT + sizeof(struct tcphdr) ||
/* If PSH is not set, packet should be
* full sized, provided peer TCP is not badly broken.
* This observation (if it is correct 8)) allows
* to handle super-low mtu links fairly.
*/
(len >= TCP_MIN_MSS + sizeof(struct tcphdr) &&
!(tcp_flag_word(tcp_hdr(skb)) & TCP_REMNANT))) {
/* Subtract also invariant (if peer is RFC compliant),
* tcp header plus fixed timestamp option length.
* Resulting "len" is MSS free of SACK jitter.
*/
len -= tcp_sk(sk)->tcp_header_len;
icsk->icsk_ack.last_seg_size = len;
if (len == lss) {
icsk->icsk_ack.rcv_mss = len;
return;
}
}
if (icsk->icsk_ack.pending & ICSK_ACK_PUSHED)
icsk->icsk_ack.pending |= ICSK_ACK_PUSHED2;
icsk->icsk_ack.pending |= ICSK_ACK_PUSHED;
}
}
static void tcp_incr_quickack(struct sock *sk)
{
struct inet_connection_sock *icsk = inet_csk(sk);
unsigned int quickacks = tcp_sk(sk)->rcv_wnd / (2 * icsk->icsk_ack.rcv_mss);
if (quickacks == 0)
quickacks = 2;
if (quickacks > icsk->icsk_ack.quick)
icsk->icsk_ack.quick = min(quickacks, TCP_MAX_QUICKACKS);
}
static void tcp_enter_quickack_mode(struct sock *sk)
{
struct inet_connection_sock *icsk = inet_csk(sk);
tcp_incr_quickack(sk);
icsk->icsk_ack.pingpong = 0;
icsk->icsk_ack.ato = TCP_ATO_MIN;
}
/* Send ACKs quickly, if "quick" count is not exhausted
* and the session is not interactive.
*/
static inline bool tcp_in_quickack_mode(const struct sock *sk)
{
const struct inet_connection_sock *icsk = inet_csk(sk);
return icsk->icsk_ack.quick && !icsk->icsk_ack.pingpong;
}
static inline void TCP_ECN_queue_cwr(struct tcp_sock *tp)
{
if (tp->ecn_flags & TCP_ECN_OK)
tp->ecn_flags |= TCP_ECN_QUEUE_CWR;
}
static inline void TCP_ECN_accept_cwr(struct tcp_sock *tp, const struct sk_buff *skb)
{
if (tcp_hdr(skb)->cwr)
tp->ecn_flags &= ~TCP_ECN_DEMAND_CWR;
}
static inline void TCP_ECN_withdraw_cwr(struct tcp_sock *tp)
{
tp->ecn_flags &= ~TCP_ECN_DEMAND_CWR;
}
static inline void TCP_ECN_check_ce(struct tcp_sock *tp, const struct sk_buff *skb)
{
if (!(tp->ecn_flags & TCP_ECN_OK))
return;
switch (TCP_SKB_CB(skb)->ip_dsfield & INET_ECN_MASK) {
case INET_ECN_NOT_ECT:
/* Funny extension: if ECT is not set on a segment,
* and we already seen ECT on a previous segment,
* it is probably a retransmit.
*/
if (tp->ecn_flags & TCP_ECN_SEEN)
tcp_enter_quickack_mode((struct sock *)tp);
break;
case INET_ECN_CE:
if (!(tp->ecn_flags & TCP_ECN_DEMAND_CWR)) {
/* Better not delay acks, sender can have a very low cwnd */
tcp_enter_quickack_mode((struct sock *)tp);
tp->ecn_flags |= TCP_ECN_DEMAND_CWR;
}
/* fallinto */
default:
tp->ecn_flags |= TCP_ECN_SEEN;
}
}
static inline void TCP_ECN_rcv_synack(struct tcp_sock *tp, const struct tcphdr *th)
{
if ((tp->ecn_flags & TCP_ECN_OK) && (!th->ece || th->cwr))
tp->ecn_flags &= ~TCP_ECN_OK;
}
static inline void TCP_ECN_rcv_syn(struct tcp_sock *tp, const struct tcphdr *th)
{
if ((tp->ecn_flags & TCP_ECN_OK) && (!th->ece || !th->cwr))
tp->ecn_flags &= ~TCP_ECN_OK;
}
static bool TCP_ECN_rcv_ecn_echo(const struct tcp_sock *tp, const struct tcphdr *th)
{
if (th->ece && !th->syn && (tp->ecn_flags & TCP_ECN_OK))
return true;
return false;
}
/* Buffer size and advertised window tuning.
*
* 1. Tuning sk->sk_sndbuf, when connection enters established state.
*/
static void tcp_sndbuf_expand(struct sock *sk)
{
const struct tcp_sock *tp = tcp_sk(sk);
int sndmem, per_mss;
u32 nr_segs;
/* Worst case is non GSO/TSO : each frame consumes one skb
* and skb->head is kmalloced using power of two area of memory
*/
per_mss = max_t(u32, tp->rx_opt.mss_clamp, tp->mss_cache) +
MAX_TCP_HEADER +
SKB_DATA_ALIGN(sizeof(struct skb_shared_info));
per_mss = roundup_pow_of_two(per_mss) +
SKB_DATA_ALIGN(sizeof(struct sk_buff));
nr_segs = max_t(u32, TCP_INIT_CWND, tp->snd_cwnd);
nr_segs = max_t(u32, nr_segs, tp->reordering + 1);
/* Fast Recovery (RFC 5681 3.2) :
* Cubic needs 1.7 factor, rounded to 2 to include
* extra cushion (application might react slowly to POLLOUT)
*/
sndmem = 2 * nr_segs * per_mss;
if (sk->sk_sndbuf < sndmem)
sk->sk_sndbuf = min(sndmem, sysctl_tcp_wmem[2]);
}
/* 2. Tuning advertised window (window_clamp, rcv_ssthresh)
*
* All tcp_full_space() is split to two parts: "network" buffer, allocated
* forward and advertised in receiver window (tp->rcv_wnd) and
* "application buffer", required to isolate scheduling/application
* latencies from network.
* window_clamp is maximal advertised window. It can be less than
* tcp_full_space(), in this case tcp_full_space() - window_clamp
* is reserved for "application" buffer. The less window_clamp is
* the smoother our behaviour from viewpoint of network, but the lower
* throughput and the higher sensitivity of the connection to losses. 8)
*
* rcv_ssthresh is more strict window_clamp used at "slow start"
* phase to predict further behaviour of this connection.
* It is used for two goals:
* - to enforce header prediction at sender, even when application
* requires some significant "application buffer". It is check #1.
* - to prevent pruning of receive queue because of misprediction
* of receiver window. Check #2.
*
* The scheme does not work when sender sends good segments opening
* window and then starts to feed us spaghetti. But it should work
* in common situations. Otherwise, we have to rely on queue collapsing.
*/
/* Slow part of check#2. */
static int __tcp_grow_window(const struct sock *sk, const struct sk_buff *skb)
{
struct tcp_sock *tp = tcp_sk(sk);
/* Optimize this! */
int truesize = tcp_win_from_space(skb->truesize) >> 1;
int window = tcp_win_from_space(sysctl_tcp_rmem[2]) >> 1;
while (tp->rcv_ssthresh <= window) {
if (truesize <= skb->len)
return 2 * inet_csk(sk)->icsk_ack.rcv_mss;
truesize >>= 1;
window >>= 1;
}
return 0;
}
static void tcp_grow_window(struct sock *sk, const struct sk_buff *skb)
{
struct tcp_sock *tp = tcp_sk(sk);
/* Check #1 */
if (tp->rcv_ssthresh < tp->window_clamp &&
(int)tp->rcv_ssthresh < tcp_space(sk) &&
!sk_under_memory_pressure(sk)) {
int incr;
/* Check #2. Increase window, if skb with such overhead
* will fit to rcvbuf in future.
*/
if (tcp_win_from_space(skb->truesize) <= skb->len)
incr = 2 * tp->advmss;
else
incr = __tcp_grow_window(sk, skb);
if (incr) {
incr = max_t(int, incr, 2 * skb->len);
tp->rcv_ssthresh = min(tp->rcv_ssthresh + incr,
tp->window_clamp);
inet_csk(sk)->icsk_ack.quick |= 1;
}
}
}
/* 3. Tuning rcvbuf, when connection enters established state. */
static void tcp_fixup_rcvbuf(struct sock *sk)
{
u32 mss = tcp_sk(sk)->advmss;
int rcvmem;
rcvmem = 2 * SKB_TRUESIZE(mss + MAX_TCP_HEADER) *
tcp_default_init_rwnd(mss);
/* Dynamic Right Sizing (DRS) has 2 to 3 RTT latency
* Allow enough cushion so that sender is not limited by our window
*/
if (sysctl_tcp_moderate_rcvbuf)
rcvmem <<= 2;
if (sk->sk_rcvbuf < rcvmem)
sk->sk_rcvbuf = min(rcvmem, sysctl_tcp_rmem[2]);
}
/* 4. Try to fixup all. It is made immediately after connection enters
* established state.
*/
void tcp_init_buffer_space(struct sock *sk)
{
struct tcp_sock *tp = tcp_sk(sk);
int maxwin;
if (!(sk->sk_userlocks & SOCK_RCVBUF_LOCK))
tcp_fixup_rcvbuf(sk);
if (!(sk->sk_userlocks & SOCK_SNDBUF_LOCK))
tcp_sndbuf_expand(sk);
tp->rcvq_space.space = tp->rcv_wnd;
tp->rcvq_space.time = tcp_time_stamp;
tp->rcvq_space.seq = tp->copied_seq;
maxwin = tcp_full_space(sk);
if (tp->window_clamp >= maxwin) {
tp->window_clamp = maxwin;
if (sysctl_tcp_app_win && maxwin > 4 * tp->advmss)
tp->window_clamp = max(maxwin -
(maxwin >> sysctl_tcp_app_win),
4 * tp->advmss);
}
/* Force reservation of one segment. */
if (sysctl_tcp_app_win &&
tp->window_clamp > 2 * tp->advmss &&
tp->window_clamp + tp->advmss > maxwin)
tp->window_clamp = max(2 * tp->advmss, maxwin - tp->advmss);
tp->rcv_ssthresh = min(tp->rcv_ssthresh, tp->window_clamp);
tp->snd_cwnd_stamp = tcp_time_stamp;
}
/* 5. Recalculate window clamp after socket hit its memory bounds. */
static void tcp_clamp_window(struct sock *sk)
{
struct tcp_sock *tp = tcp_sk(sk);
struct inet_connection_sock *icsk = inet_csk(sk);
icsk->icsk_ack.quick = 0;
if (sk->sk_rcvbuf < sysctl_tcp_rmem[2] &&
!(sk->sk_userlocks & SOCK_RCVBUF_LOCK) &&
!sk_under_memory_pressure(sk) &&
sk_memory_allocated(sk) < sk_prot_mem_limits(sk, 0)) {
sk->sk_rcvbuf = min(atomic_read(&sk->sk_rmem_alloc),
sysctl_tcp_rmem[2]);
}
if (atomic_read(&sk->sk_rmem_alloc) > sk->sk_rcvbuf)
tp->rcv_ssthresh = min(tp->window_clamp, 2U * tp->advmss);
}
/* Initialize RCV_MSS value.
* RCV_MSS is an our guess about MSS used by the peer.
* We haven't any direct information about the MSS.
* It's better to underestimate the RCV_MSS rather than overestimate.
* Overestimations make us ACKing less frequently than needed.
* Underestimations are more easy to detect and fix by tcp_measure_rcv_mss().
*/
void tcp_initialize_rcv_mss(struct sock *sk)
{
const struct tcp_sock *tp = tcp_sk(sk);
unsigned int hint = min_t(unsigned int, tp->advmss, tp->mss_cache);
hint = min(hint, tp->rcv_wnd / 2);
hint = min(hint, TCP_MSS_DEFAULT);
hint = max(hint, TCP_MIN_MSS);
inet_csk(sk)->icsk_ack.rcv_mss = hint;
}
EXPORT_SYMBOL(tcp_initialize_rcv_mss);
/* Receiver "autotuning" code.
*
* The algorithm for RTT estimation w/o timestamps is based on
* Dynamic Right-Sizing (DRS) by Wu Feng and Mike Fisk of LANL.
* <http://public.lanl.gov/radiant/pubs.html#DRS>
*
* More detail on this code can be found at
* <http://staff.psc.edu/jheffner/>,
* though this reference is out of date. A new paper
* is pending.
*/
static void tcp_rcv_rtt_update(struct tcp_sock *tp, u32 sample, int win_dep)
{
u32 new_sample = tp->rcv_rtt_est.rtt;
long m = sample;
if (m == 0)
m = 1;
if (new_sample != 0) {
/* If we sample in larger samples in the non-timestamp
* case, we could grossly overestimate the RTT especially
* with chatty applications or bulk transfer apps which
* are stalled on filesystem I/O.
*
* Also, since we are only going for a minimum in the
* non-timestamp case, we do not smooth things out
* else with timestamps disabled convergence takes too
* long.
*/
if (!win_dep) {
m -= (new_sample >> 3);
new_sample += m;
} else {
m <<= 3;
if (m < new_sample)
new_sample = m;
}
} else {
/* No previous measure. */
new_sample = m << 3;
}
if (tp->rcv_rtt_est.rtt != new_sample)
tp->rcv_rtt_est.rtt = new_sample;
}
static inline void tcp_rcv_rtt_measure(struct tcp_sock *tp)
{
if (tp->rcv_rtt_est.time == 0)
goto new_measure;
if (before(tp->rcv_nxt, tp->rcv_rtt_est.seq))
return;
tcp_rcv_rtt_update(tp, tcp_time_stamp - tp->rcv_rtt_est.time, 1);
new_measure:
tp->rcv_rtt_est.seq = tp->rcv_nxt + tp->rcv_wnd;
tp->rcv_rtt_est.time = tcp_time_stamp;
}
static inline void tcp_rcv_rtt_measure_ts(struct sock *sk,
const struct sk_buff *skb)
{
struct tcp_sock *tp = tcp_sk(sk);
if (tp->rx_opt.rcv_tsecr &&
(TCP_SKB_CB(skb)->end_seq -
TCP_SKB_CB(skb)->seq >= inet_csk(sk)->icsk_ack.rcv_mss))
tcp_rcv_rtt_update(tp, tcp_time_stamp - tp->rx_opt.rcv_tsecr, 0);
}
/*
* This function should be called every time data is copied to user space.
* It calculates the appropriate TCP receive buffer space.
*/
void tcp_rcv_space_adjust(struct sock *sk)
{
struct tcp_sock *tp = tcp_sk(sk);
int time;
int copied;
time = tcp_time_stamp - tp->rcvq_space.time;
if (time < (tp->rcv_rtt_est.rtt >> 3) || tp->rcv_rtt_est.rtt == 0)
return;
/* Number of bytes copied to user in last RTT */
copied = tp->copied_seq - tp->rcvq_space.seq;
if (copied <= tp->rcvq_space.space)
goto new_measure;
/* A bit of theory :
* copied = bytes received in previous RTT, our base window
* To cope with packet losses, we need a 2x factor
* To cope with slow start, and sender growing its cwin by 100 %
* every RTT, we need a 4x factor, because the ACK we are sending
* now is for the next RTT, not the current one :
* <prev RTT . ><current RTT .. ><next RTT .... >
*/
if (sysctl_tcp_moderate_rcvbuf &&
!(sk->sk_userlocks & SOCK_RCVBUF_LOCK)) {
int rcvwin, rcvmem, rcvbuf;
/* minimal window to cope with packet losses, assuming
* steady state. Add some cushion because of small variations.
*/
rcvwin = (copied << 1) + 16 * tp->advmss;
/* If rate increased by 25%,
* assume slow start, rcvwin = 3 * copied
* If rate increased by 50%,
* assume sender can use 2x growth, rcvwin = 4 * copied
*/
if (copied >=
tp->rcvq_space.space + (tp->rcvq_space.space >> 2)) {
if (copied >=
tp->rcvq_space.space + (tp->rcvq_space.space >> 1))
rcvwin <<= 1;
else
rcvwin += (rcvwin >> 1);
}
rcvmem = SKB_TRUESIZE(tp->advmss + MAX_TCP_HEADER);
while (tcp_win_from_space(rcvmem) < tp->advmss)
rcvmem += 128;
rcvbuf = min(rcvwin / tp->advmss * rcvmem, sysctl_tcp_rmem[2]);
if (rcvbuf > sk->sk_rcvbuf) {
sk->sk_rcvbuf = rcvbuf;
/* Make the window clamp follow along. */
tp->window_clamp = rcvwin;
}
}
tp->rcvq_space.space = copied;
new_measure:
tp->rcvq_space.seq = tp->copied_seq;
tp->rcvq_space.time = tcp_time_stamp;
}
/* There is something which you must keep in mind when you analyze the
* behavior of the tp->ato delayed ack timeout interval. When a
* connection starts up, we want to ack as quickly as possible. The
* problem is that "good" TCP's do slow start at the beginning of data
* transmission. The means that until we send the first few ACK's the
* sender will sit on his end and only queue most of his data, because
* he can only send snd_cwnd unacked packets at any given time. For
* each ACK we send, he increments snd_cwnd and transmits more of his
* queue. -DaveM
*/
static void tcp_event_data_recv(struct sock *sk, struct sk_buff *skb)
{
struct tcp_sock *tp = tcp_sk(sk);
struct inet_connection_sock *icsk = inet_csk(sk);
u32 now;
inet_csk_schedule_ack(sk);
tcp_measure_rcv_mss(sk, skb);
tcp_rcv_rtt_measure(tp);
now = tcp_time_stamp;
if (!icsk->icsk_ack.ato) {
/* The _first_ data packet received, initialize
* delayed ACK engine.
*/
tcp_incr_quickack(sk);
icsk->icsk_ack.ato = TCP_ATO_MIN;
} else {
int m = now - icsk->icsk_ack.lrcvtime;
if (m <= TCP_ATO_MIN / 2) {
/* The fastest case is the first. */
icsk->icsk_ack.ato = (icsk->icsk_ack.ato >> 1) + TCP_ATO_MIN / 2;
} else if (m < icsk->icsk_ack.ato) {
icsk->icsk_ack.ato = (icsk->icsk_ack.ato >> 1) + m;
if (icsk->icsk_ack.ato > icsk->icsk_rto)
icsk->icsk_ack.ato = icsk->icsk_rto;
} else if (m > icsk->icsk_rto) {
/* Too long gap. Apparently sender failed to
* restart window, so that we send ACKs quickly.
*/
tcp_incr_quickack(sk);
sk_mem_reclaim(sk);
}
}
icsk->icsk_ack.lrcvtime = now;
TCP_ECN_check_ce(tp, skb);
if (skb->len >= 128)
tcp_grow_window(sk, skb);
}
/* Called to compute a smoothed rtt estimate. The data fed to this
* routine either comes from timestamps, or from segments that were
* known _not_ to have been retransmitted [see Karn/Partridge
* Proceedings SIGCOMM 87]. The algorithm is from the SIGCOMM 88
* piece by Van Jacobson.
* NOTE: the next three routines used to be one big routine.
* To save cycles in the RFC 1323 implementation it was better to break
* it up into three procedures. -- erics
*/
static void tcp_rtt_estimator(struct sock *sk, const __u32 mrtt)
{
struct tcp_sock *tp = tcp_sk(sk);
long m = mrtt; /* RTT */
u32 srtt = tp->srtt;
/* The following amusing code comes from Jacobson's
* article in SIGCOMM '88. Note that rtt and mdev
* are scaled versions of rtt and mean deviation.
* This is designed to be as fast as possible
* m stands for "measurement".
*
* On a 1990 paper the rto value is changed to:
* RTO = rtt + 4 * mdev
*
* Funny. This algorithm seems to be very broken.
* These formulae increase RTO, when it should be decreased, increase
* too slowly, when it should be increased quickly, decrease too quickly
* etc. I guess in BSD RTO takes ONE value, so that it is absolutely
* does not matter how to _calculate_ it. Seems, it was trap
* that VJ failed to avoid. 8)
*/
if (srtt != 0) {
m -= (srtt >> 3); /* m is now error in rtt est */
srtt += m; /* rtt = 7/8 rtt + 1/8 new */
if (m < 0) {
m = -m; /* m is now abs(error) */
m -= (tp->mdev >> 2); /* similar update on mdev */
/* This is similar to one of Eifel findings.
* Eifel blocks mdev updates when rtt decreases.
* This solution is a bit different: we use finer gain
* for mdev in this case (alpha*beta).
* Like Eifel it also prevents growth of rto,
* but also it limits too fast rto decreases,
* happening in pure Eifel.
*/
if (m > 0)
m >>= 3;
} else {
m -= (tp->mdev >> 2); /* similar update on mdev */
}
tp->mdev += m; /* mdev = 3/4 mdev + 1/4 new */
if (tp->mdev > tp->mdev_max) {
tp->mdev_max = tp->mdev;
if (tp->mdev_max > tp->rttvar)
tp->rttvar = tp->mdev_max;
}
if (after(tp->snd_una, tp->rtt_seq)) {
if (tp->mdev_max < tp->rttvar)
tp->rttvar -= (tp->rttvar - tp->mdev_max) >> 2;
tp->rtt_seq = tp->snd_nxt;
tp->mdev_max = tcp_rto_min(sk);
}
} else {
/* no previous measure. */
srtt = m << 3; /* take the measured time to be rtt */
tp->mdev = m << 1; /* make sure rto = 3*rtt */
tp->mdev_max = tp->rttvar = max(tp->mdev, tcp_rto_min(sk));
tp->rtt_seq = tp->snd_nxt;
}
tp->srtt = max(1U, srtt);
}
/* Set the sk_pacing_rate to allow proper sizing of TSO packets.
* Note: TCP stack does not yet implement pacing.
* FQ packet scheduler can be used to implement cheap but effective
* TCP pacing, to smooth the burst on large writes when packets
* in flight is significantly lower than cwnd (or rwin)
*/
static void tcp_update_pacing_rate(struct sock *sk)
{
const struct tcp_sock *tp = tcp_sk(sk);
u64 rate;
/* set sk_pacing_rate to 200 % of current rate (mss * cwnd / srtt) */
rate = (u64)tp->mss_cache * 2 * (HZ << 3);
rate *= max(tp->snd_cwnd, tp->packets_out);
/* Correction for small srtt and scheduling constraints.
* For small rtt, consider noise is too high, and use
* the minimal value (srtt = 1 -> 125 us for HZ=1000)
*
* We probably need usec resolution in the future.
* Note: This also takes care of possible srtt=0 case,
* when tcp_rtt_estimator() was not yet called.
*/
if (tp->srtt > 8 + 2)
do_div(rate, tp->srtt);
/* ACCESS_ONCE() is needed because sch_fq fetches sk_pacing_rate
* without any lock. We want to make sure compiler wont store
* intermediate values in this location.
*/
ACCESS_ONCE(sk->sk_pacing_rate) = min_t(u64, rate,
sk->sk_max_pacing_rate);
}
/* Calculate rto without backoff. This is the second half of Van Jacobson's
* routine referred to above.
*/
static void tcp_set_rto(struct sock *sk)
{
const struct tcp_sock *tp = tcp_sk(sk);
/* Old crap is replaced with new one. 8)
*
* More seriously:
* 1. If rtt variance happened to be less 50msec, it is hallucination.
* It cannot be less due to utterly erratic ACK generation made
* at least by solaris and freebsd. "Erratic ACKs" has _nothing_
* to do with delayed acks, because at cwnd>2 true delack timeout
* is invisible. Actually, Linux-2.4 also generates erratic
* ACKs in some circumstances.
*/
inet_csk(sk)->icsk_rto = __tcp_set_rto(tp);
/* 2. Fixups made earlier cannot be right.
* If we do not estimate RTO correctly without them,
* all the algo is pure shit and should be replaced
* with correct one. It is exactly, which we pretend to do.
*/
/* NOTE: clamping at TCP_RTO_MIN is not required, current algo
* guarantees that rto is higher.
*/
tcp_bound_rto(sk);
}
__u32 tcp_init_cwnd(const struct tcp_sock *tp, const struct dst_entry *dst)
{
__u32 cwnd = (dst ? dst_metric(dst, RTAX_INITCWND) : 0);
if (!cwnd)
cwnd = TCP_INIT_CWND;
return min_t(__u32, cwnd, tp->snd_cwnd_clamp);
}
/*
* Packet counting of FACK is based on in-order assumptions, therefore TCP
* disables it when reordering is detected
*/
void tcp_disable_fack(struct tcp_sock *tp)
{
/* RFC3517 uses different metric in lost marker => reset on change */
if (tcp_is_fack(tp))
tp->lost_skb_hint = NULL;
tp->rx_opt.sack_ok &= ~TCP_FACK_ENABLED;
}
/* Take a notice that peer is sending D-SACKs */
static void tcp_dsack_seen(struct tcp_sock *tp)
{
tp->rx_opt.sack_ok |= TCP_DSACK_SEEN;
}
static void tcp_update_reordering(struct sock *sk, const int metric,
const int ts)
{
struct tcp_sock *tp = tcp_sk(sk);
if (metric > tp->reordering) {
int mib_idx;
tp->reordering = min(TCP_MAX_REORDERING, metric);
/* This exciting event is worth to be remembered. 8) */
if (ts)
mib_idx = LINUX_MIB_TCPTSREORDER;
else if (tcp_is_reno(tp))
mib_idx = LINUX_MIB_TCPRENOREORDER;
else if (tcp_is_fack(tp))
mib_idx = LINUX_MIB_TCPFACKREORDER;
else
mib_idx = LINUX_MIB_TCPSACKREORDER;
NET_INC_STATS_BH(sock_net(sk), mib_idx);
#if FASTRETRANS_DEBUG > 1
pr_debug("Disorder%d %d %u f%u s%u rr%d\n",
tp->rx_opt.sack_ok, inet_csk(sk)->icsk_ca_state,
tp->reordering,
tp->fackets_out,
tp->sacked_out,
tp->undo_marker ? tp->undo_retrans : 0);
#endif
tcp_disable_fack(tp);
}
if (metric > 0)
tcp_disable_early_retrans(tp);
}
/* This must be called before lost_out is incremented */
static void tcp_verify_retransmit_hint(struct tcp_sock *tp, struct sk_buff *skb)
{
if ((tp->retransmit_skb_hint == NULL) ||
before(TCP_SKB_CB(skb)->seq,
TCP_SKB_CB(tp->retransmit_skb_hint)->seq))
tp->retransmit_skb_hint = skb;
if (!tp->lost_out ||
after(TCP_SKB_CB(skb)->end_seq, tp->retransmit_high))
tp->retransmit_high = TCP_SKB_CB(skb)->end_seq;
}
static void tcp_skb_mark_lost(struct tcp_sock *tp, struct sk_buff *skb)
{
if (!(TCP_SKB_CB(skb)->sacked & (TCPCB_LOST|TCPCB_SACKED_ACKED))) {
tcp_verify_retransmit_hint(tp, skb);
tp->lost_out += tcp_skb_pcount(skb);
TCP_SKB_CB(skb)->sacked |= TCPCB_LOST;
}
}
static void tcp_skb_mark_lost_uncond_verify(struct tcp_sock *tp,
struct sk_buff *skb)
{
tcp_verify_retransmit_hint(tp, skb);
if (!(TCP_SKB_CB(skb)->sacked & (TCPCB_LOST|TCPCB_SACKED_ACKED))) {
tp->lost_out += tcp_skb_pcount(skb);
TCP_SKB_CB(skb)->sacked |= TCPCB_LOST;
}
}
/* This procedure tags the retransmission queue when SACKs arrive.
*
* We have three tag bits: SACKED(S), RETRANS(R) and LOST(L).
* Packets in queue with these bits set are counted in variables
* sacked_out, retrans_out and lost_out, correspondingly.
*
* Valid combinations are:
* Tag InFlight Description
* 0 1 - orig segment is in flight.
* S 0 - nothing flies, orig reached receiver.
* L 0 - nothing flies, orig lost by net.
* R 2 - both orig and retransmit are in flight.
* L|R 1 - orig is lost, retransmit is in flight.
* S|R 1 - orig reached receiver, retrans is still in flight.
* (L|S|R is logically valid, it could occur when L|R is sacked,
* but it is equivalent to plain S and code short-curcuits it to S.
* L|S is logically invalid, it would mean -1 packet in flight 8))
*
* These 6 states form finite state machine, controlled by the following events:
* 1. New ACK (+SACK) arrives. (tcp_sacktag_write_queue())
* 2. Retransmission. (tcp_retransmit_skb(), tcp_xmit_retransmit_queue())
* 3. Loss detection event of two flavors:
* A. Scoreboard estimator decided the packet is lost.
* A'. Reno "three dupacks" marks head of queue lost.
* A''. Its FACK modification, head until snd.fack is lost.
* B. SACK arrives sacking SND.NXT at the moment, when the
* segment was retransmitted.
* 4. D-SACK added new rule: D-SACK changes any tag to S.
*
* It is pleasant to note, that state diagram turns out to be commutative,
* so that we are allowed not to be bothered by order of our actions,
* when multiple events arrive simultaneously. (see the function below).
*
* Reordering detection.
* --------------------
* Reordering metric is maximal distance, which a packet can be displaced
* in packet stream. With SACKs we can estimate it:
*
* 1. SACK fills old hole and the corresponding segment was not
* ever retransmitted -> reordering. Alas, we cannot use it
* when segment was retransmitted.
* 2. The last flaw is solved with D-SACK. D-SACK arrives
* for retransmitted and already SACKed segment -> reordering..
* Both of these heuristics are not used in Loss state, when we cannot
* account for retransmits accurately.
*
* SACK block validation.
* ----------------------
*
* SACK block range validation checks that the received SACK block fits to
* the expected sequence limits, i.e., it is between SND.UNA and SND.NXT.
* Note that SND.UNA is not included to the range though being valid because
* it means that the receiver is rather inconsistent with itself reporting
* SACK reneging when it should advance SND.UNA. Such SACK block this is
* perfectly valid, however, in light of RFC2018 which explicitly states
* that "SACK block MUST reflect the newest segment. Even if the newest
* segment is going to be discarded ...", not that it looks very clever
* in case of head skb. Due to potentional receiver driven attacks, we
* choose to avoid immediate execution of a walk in write queue due to
* reneging and defer head skb's loss recovery to standard loss recovery
* procedure that will eventually trigger (nothing forbids us doing this).
*
* Implements also blockage to start_seq wrap-around. Problem lies in the
* fact that though start_seq (s) is before end_seq (i.e., not reversed),
* there's no guarantee that it will be before snd_nxt (n). The problem
* happens when start_seq resides between end_seq wrap (e_w) and snd_nxt
* wrap (s_w):
*
* <- outs wnd -> <- wrapzone ->
* u e n u_w e_w s n_w
* | | | | | | |
* |<------------+------+----- TCP seqno space --------------+---------->|
* ...-- <2^31 ->| |<--------...
* ...---- >2^31 ------>| |<--------...
*
* Current code wouldn't be vulnerable but it's better still to discard such
* crazy SACK blocks. Doing this check for start_seq alone closes somewhat
* similar case (end_seq after snd_nxt wrap) as earlier reversed check in
* snd_nxt wrap -> snd_una region will then become "well defined", i.e.,
* equal to the ideal case (infinite seqno space without wrap caused issues).
*
* With D-SACK the lower bound is extended to cover sequence space below
* SND.UNA down to undo_marker, which is the last point of interest. Yet
* again, D-SACK block must not to go across snd_una (for the same reason as
* for the normal SACK blocks, explained above). But there all simplicity
* ends, TCP might receive valid D-SACKs below that. As long as they reside
* fully below undo_marker they do not affect behavior in anyway and can
* therefore be safely ignored. In rare cases (which are more or less
* theoretical ones), the D-SACK will nicely cross that boundary due to skb
* fragmentation and packet reordering past skb's retransmission. To consider
* them correctly, the acceptable range must be extended even more though
* the exact amount is rather hard to quantify. However, tp->max_window can
* be used as an exaggerated estimate.
*/
static bool tcp_is_sackblock_valid(struct tcp_sock *tp, bool is_dsack,
u32 start_seq, u32 end_seq)
{
/* Too far in future, or reversed (interpretation is ambiguous) */
if (after(end_seq, tp->snd_nxt) || !before(start_seq, end_seq))
return false;
/* Nasty start_seq wrap-around check (see comments above) */
if (!before(start_seq, tp->snd_nxt))
return false;
/* In outstanding window? ...This is valid exit for D-SACKs too.
* start_seq == snd_una is non-sensical (see comments above)