linux_dsm_epyc7002/include/linux/tcp.h

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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.
*
* Definitions for the TCP protocol.
*
* Version: @(#)tcp.h 1.0.2 04/28/93
*
* Author: Fred N. van Kempen, <waltje@uWalt.NL.Mugnet.ORG>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version
* 2 of the License, or (at your option) any later version.
*/
#ifndef _LINUX_TCP_H
#define _LINUX_TCP_H
#include <linux/skbuff.h>
#include <linux/win_minmax.h>
#include <net/sock.h>
#include <net/inet_connection_sock.h>
#include <net/inet_timewait_sock.h>
#include <uapi/linux/tcp.h>
static inline struct tcphdr *tcp_hdr(const struct sk_buff *skb)
{
return (struct tcphdr *)skb_transport_header(skb);
}
static inline unsigned int __tcp_hdrlen(const struct tcphdr *th)
{
return th->doff * 4;
}
static inline unsigned int tcp_hdrlen(const struct sk_buff *skb)
{
return __tcp_hdrlen(tcp_hdr(skb));
}
static inline struct tcphdr *inner_tcp_hdr(const struct sk_buff *skb)
{
return (struct tcphdr *)skb_inner_transport_header(skb);
}
static inline unsigned int inner_tcp_hdrlen(const struct sk_buff *skb)
{
return inner_tcp_hdr(skb)->doff * 4;
}
static inline unsigned int tcp_optlen(const struct sk_buff *skb)
{
return (tcp_hdr(skb)->doff - 5) * 4;
}
/* TCP Fast Open */
#define TCP_FASTOPEN_COOKIE_MIN 4 /* Min Fast Open Cookie size in bytes */
#define TCP_FASTOPEN_COOKIE_MAX 16 /* Max Fast Open Cookie size in bytes */
#define TCP_FASTOPEN_COOKIE_SIZE 8 /* the size employed by this impl. */
/* TCP Fast Open Cookie as stored in memory */
struct tcp_fastopen_cookie {
union {
u8 val[TCP_FASTOPEN_COOKIE_MAX];
#if IS_ENABLED(CONFIG_IPV6)
struct in6_addr addr;
#endif
};
s8 len;
bool exp; /* In RFC6994 experimental option format */
};
/* This defines a selective acknowledgement block. */
struct tcp_sack_block_wire {
__be32 start_seq;
__be32 end_seq;
};
struct tcp_sack_block {
u32 start_seq;
u32 end_seq;
};
/*These are used to set the sack_ok field in struct tcp_options_received */
#define TCP_SACK_SEEN (1 << 0) /*1 = peer is SACK capable, */
#define TCP_DSACK_SEEN (1 << 2) /*1 = DSACK was received from peer*/
struct tcp_options_received {
/* PAWS/RTTM data */
long ts_recent_stamp;/* Time we stored ts_recent (for aging) */
u32 ts_recent; /* Time stamp to echo next */
u32 rcv_tsval; /* Time stamp value */
u32 rcv_tsecr; /* Time stamp echo reply */
u16 saw_tstamp : 1, /* Saw TIMESTAMP on last packet */
tstamp_ok : 1, /* TIMESTAMP seen on SYN packet */
dsack : 1, /* D-SACK is scheduled */
wscale_ok : 1, /* Wscale seen on SYN packet */
sack_ok : 3, /* SACK seen on SYN packet */
smc_ok : 1, /* SMC seen on SYN packet */
snd_wscale : 4, /* Window scaling received from sender */
rcv_wscale : 4; /* Window scaling to send to receiver */
u8 num_sacks; /* Number of SACK blocks */
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 01:17:05 +07:00
u16 user_mss; /* mss requested by user in ioctl */
u16 mss_clamp; /* Maximal mss, negotiated at connection setup */
};
static inline void tcp_clear_options(struct tcp_options_received *rx_opt)
{
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 01:17:05 +07:00
rx_opt->tstamp_ok = rx_opt->sack_ok = 0;
rx_opt->wscale_ok = rx_opt->snd_wscale = 0;
#if IS_ENABLED(CONFIG_SMC)
rx_opt->smc_ok = 0;
#endif
}
/* This is the max number of SACKS that we'll generate and process. It's safe
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 01:17:05 +07:00
* to increase this, although since:
* size = TCPOLEN_SACK_BASE_ALIGNED (4) + n * TCPOLEN_SACK_PERBLOCK (8)
* only four options will fit in a standard TCP header */
#define TCP_NUM_SACKS 4
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 01:17:05 +07:00
struct tcp_request_sock_ops;
struct tcp_request_sock {
struct inet_request_sock req;
const struct tcp_request_sock_ops *af_specific;
u64 snt_synack; /* first SYNACK sent time */
bool tfo_listener;
u32 txhash;
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 01:17:05 +07:00
u32 rcv_isn;
u32 snt_isn;
u32 ts_off;
u32 last_oow_ack_time; /* last SYNACK */
u32 rcv_nxt; /* the ack # by SYNACK. For
* FastOpen it's the seq#
* after data-in-SYN.
*/
};
static inline struct tcp_request_sock *tcp_rsk(const struct request_sock *req)
{
return (struct tcp_request_sock *)req;
}
struct tcp_sock {
/* inet_connection_sock has to be the first member of tcp_sock */
struct inet_connection_sock inet_conn;
u16 tcp_header_len; /* Bytes of tcp header to send */
tcp: refine TSO autosizing Commit 95bd09eb2750 ("tcp: TSO packets automatic sizing") tried to control TSO size, but did this at the wrong place (sendmsg() time) At sendmsg() time, we might have a pessimistic view of flow rate, and we end up building very small skbs (with 2 MSS per skb). This is bad because : - It sends small TSO packets even in Slow Start where rate quickly increases. - It tends to make socket write queue very big, increasing tcp_ack() processing time, but also increasing memory needs, not necessarily accounted for, as fast clones overhead is currently ignored. - Lower GRO efficiency and more ACK packets. Servers with a lot of small lived connections suffer from this. Lets instead fill skbs as much as possible (64KB of payload), but split them at xmit time, when we have a precise idea of the flow rate. skb split is actually quite efficient. Patch looks bigger than necessary, because TCP Small Queue decision now has to take place after the eventual split. As Neal suggested, introduce a new tcp_tso_autosize() helper, so that tcp_tso_should_defer() can be synchronized on same goal. Rename tp->xmit_size_goal_segs to tp->gso_segs, as this variable contains number of mss that we can put in GSO packet, and is not related to the autosizing goal anymore. Tested: 40 ms rtt link nstat >/dev/null netperf -H remote -l -2000000 -- -s 1000000 nstat | egrep "IpInReceives|IpOutRequests|TcpOutSegs|IpExtOutOctets" Before patch : Recv Send Send Socket Socket Message Elapsed Size Size Size Time Throughput bytes bytes bytes secs. 10^6bits/s 87380 2000000 2000000 0.36 44.22 IpInReceives 600 0.0 IpOutRequests 599 0.0 TcpOutSegs 1397 0.0 IpExtOutOctets 2033249 0.0 After patch : Recv Send Send Socket Socket Message Elapsed Size Size Size Time Throughput bytes bytes bytes secs. 10^6bits/sec 87380 2000000 2000000 0.36 44.27 IpInReceives 221 0.0 IpOutRequests 232 0.0 TcpOutSegs 1397 0.0 IpExtOutOctets 2013953 0.0 Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Acked-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-12-08 03:22:18 +07:00
u16 gso_segs; /* Max number of segs per GSO packet */
/*
* Header prediction flags
* 0x5?10 << 16 + snd_wnd in net byte order
*/
__be32 pred_flags;
/*
* RFC793 variables by their proper names. This means you can
* read the code and the spec side by side (and laugh ...)
* See RFC793 and RFC1122. The RFC writes these in capitals.
*/
u64 bytes_received; /* RFC4898 tcpEStatsAppHCThruOctetsReceived
* sum(delta(rcv_nxt)), or how many bytes
* were acked.
*/
u32 segs_in; /* RFC4898 tcpEStatsPerfSegsIn
* total number of segments in.
*/
tcp: Add RFC4898 tcpEStatsPerfDataSegsOut/In Per RFC4898, they count segments sent/received containing a positive length data segment (that includes retransmission segments carrying data). Unlike tcpi_segs_out/in, tcpi_data_segs_out/in excludes segments carrying no data (e.g. pure ack). The patch also updates the segs_in in tcp_fastopen_add_skb() so that segs_in >= data_segs_in property is kept. Together with retransmission data, tcpi_data_segs_out gives a better signal on the rxmit rate. v6: Rebase on the latest net-next v5: Eric pointed out that checking skb->len is still needed in tcp_fastopen_add_skb() because skb can carry a FIN without data. Hence, instead of open coding segs_in and data_segs_in, tcp_segs_in() helper is used. Comment is added to the fastopen case to explain why segs_in has to be reset and tcp_segs_in() has to be called before __skb_pull(). v4: Add comment to the changes in tcp_fastopen_add_skb() and also add remark on this case in the commit message. v3: Add const modifier to the skb parameter in tcp_segs_in() v2: Rework based on recent fix by Eric: commit a9d99ce28ed3 ("tcp: fix tcpi_segs_in after connection establishment") Signed-off-by: Martin KaFai Lau <kafai@fb.com> Cc: Chris Rapier <rapier@psc.edu> Cc: Eric Dumazet <edumazet@google.com> Cc: Marcelo Ricardo Leitner <mleitner@redhat.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-03-15 00:52:15 +07:00
u32 data_segs_in; /* RFC4898 tcpEStatsPerfDataSegsIn
* total number of data segments in.
*/
u32 rcv_nxt; /* What we want to receive next */
u32 copied_seq; /* Head of yet unread data */
u32 rcv_wup; /* rcv_nxt on last window update sent */
u32 snd_nxt; /* Next sequence we send */
u32 segs_out; /* RFC4898 tcpEStatsPerfSegsOut
* The total number of segments sent.
*/
tcp: Add RFC4898 tcpEStatsPerfDataSegsOut/In Per RFC4898, they count segments sent/received containing a positive length data segment (that includes retransmission segments carrying data). Unlike tcpi_segs_out/in, tcpi_data_segs_out/in excludes segments carrying no data (e.g. pure ack). The patch also updates the segs_in in tcp_fastopen_add_skb() so that segs_in >= data_segs_in property is kept. Together with retransmission data, tcpi_data_segs_out gives a better signal on the rxmit rate. v6: Rebase on the latest net-next v5: Eric pointed out that checking skb->len is still needed in tcp_fastopen_add_skb() because skb can carry a FIN without data. Hence, instead of open coding segs_in and data_segs_in, tcp_segs_in() helper is used. Comment is added to the fastopen case to explain why segs_in has to be reset and tcp_segs_in() has to be called before __skb_pull(). v4: Add comment to the changes in tcp_fastopen_add_skb() and also add remark on this case in the commit message. v3: Add const modifier to the skb parameter in tcp_segs_in() v2: Rework based on recent fix by Eric: commit a9d99ce28ed3 ("tcp: fix tcpi_segs_in after connection establishment") Signed-off-by: Martin KaFai Lau <kafai@fb.com> Cc: Chris Rapier <rapier@psc.edu> Cc: Eric Dumazet <edumazet@google.com> Cc: Marcelo Ricardo Leitner <mleitner@redhat.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-03-15 00:52:15 +07:00
u32 data_segs_out; /* RFC4898 tcpEStatsPerfDataSegsOut
* total number of data segments sent.
*/
u64 bytes_acked; /* RFC4898 tcpEStatsAppHCThruOctetsAcked
* sum(delta(snd_una)), or how many bytes
* were acked.
*/
u32 snd_una; /* First byte we want an ack for */
u32 snd_sml; /* Last byte of the most recently transmitted small packet */
u32 rcv_tstamp; /* timestamp of last received ACK (for keepalives) */
u32 lsndtime; /* timestamp of last sent data packet (for restart window) */
u32 last_oow_ack_time; /* timestamp of last out-of-window ACK */
u32 tsoffset; /* timestamp offset */
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 12:50:31 +07:00
struct list_head tsq_node; /* anchor in tsq_tasklet.head list */
tcp: new list for sent but unacked skbs for RACK recovery This patch adds a new queue (list) that tracks the sent but not yet acked or SACKed skbs for a TCP connection. The list is chronologically ordered by skb->skb_mstamp (the head is the oldest sent skb). This list will be used to optimize TCP Rack recovery, which checks an skb's timestamp to judge if it has been lost and needs to be retransmitted. Since TCP write queue is ordered by sequence instead of sent time, RACK has to scan over the write queue to catch all eligible packets to detect lost retransmission, and iterates through SACKed skbs repeatedly. Special cares for rare events: 1. TCP repair fakes skb transmission so the send queue needs adjusted 2. SACK reneging would require re-inserting SACKed skbs into the send queue. For now I believe it's not worth the complexity to make RACK work perfectly on SACK reneging, so we do nothing here. 3. Fast Open: currently for non-TFO, send-queue correctly queues the pure SYN packet. For TFO which queues a pure SYN and then a data packet, send-queue only queues the data packet but not the pure SYN due to the structure of TFO code. This is okay because the SYN receiver would never respond with a SACK on a missing SYN (i.e. SYN is never fast-retransmitted by SACK/RACK). In order to not grow sk_buff, we use an union for the new list and _skb_refdst/destructor fields. This is a bit complicated because we need to make sure _skb_refdst and destructor are properly zeroed before skb is cloned/copied at transmit, and before being freed. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-10-05 02:59:58 +07:00
struct list_head tsorted_sent_queue; /* time-sorted sent but un-SACKed skbs */
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 12:50:31 +07:00
u32 snd_wl1; /* Sequence for window update */
u32 snd_wnd; /* The window we expect to receive */
u32 max_window; /* Maximal window ever seen from peer */
u32 mss_cache; /* Cached effective mss, not including SACKS */
u32 window_clamp; /* Maximal window to advertise */
u32 rcv_ssthresh; /* Current window clamp */
tcp: track the packet timings in RACK This patch is the first half of the RACK loss recovery. RACK loss recovery uses the notion of time instead of packet sequence (FACK) or counts (dupthresh). It's inspired by the previous FACK heuristic in tcp_mark_lost_retrans(): when a limited transmit (new data packet) is sacked, then current retransmitted sequence below the newly sacked sequence must been lost, since at least one round trip time has elapsed. But it has several limitations: 1) can't detect tail drops since it depends on limited transmit 2) is disabled upon reordering (assumes no reordering) 3) only enabled in fast recovery ut not timeout recovery RACK (Recently ACK) addresses these limitations with the notion of time instead: a packet P1 is lost if a later packet P2 is s/acked, as at least one round trip has passed. Since RACK cares about the time sequence instead of the data sequence of packets, it can detect tail drops when later retransmission is s/acked while FACK or dupthresh can't. For reordering RACK uses a dynamically adjusted reordering window ("reo_wnd") to reduce false positives on ever (small) degree of reordering. This patch implements tcp_advanced_rack() which tracks the most recent transmission time among the packets that have been delivered (ACKed or SACKed) in tp->rack.mstamp. This timestamp is the key to determine which packet has been lost. Consider an example that the sender sends six packets: T1: P1 (lost) T2: P2 T3: P3 T4: P4 T100: sack of P2. rack.mstamp = T2 T101: retransmit P1 T102: sack of P2,P3,P4. rack.mstamp = T4 T205: ACK of P4 since the hole is repaired. rack.mstamp = T101 We need to be careful about spurious retransmission because it may falsely advance tp->rack.mstamp by an RTT or an RTO, causing RACK to falsely mark all packets lost, just like a spurious timeout. We identify spurious retransmission by the ACK's TS echo value. If TS option is not applicable but the retransmission is acknowledged less than min-RTT ago, it is likely to be spurious. We refrain from using the transmission time of these spurious retransmissions. The second half is implemented in the next patch that marks packet lost using RACK timestamp. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-10-17 11:57:46 +07:00
/* Information of the most recently (s)acked skb */
struct tcp_rack {
u64 mstamp; /* (Re)sent time of the skb */
u32 rtt_us; /* Associated RTT */
u32 end_seq; /* Ending TCP sequence of the skb */
tcp: higher throughput under reordering with adaptive RACK reordering wnd Currently TCP RACK loss detection does not work well if packets are being reordered beyond its static reordering window (min_rtt/4).Under such reordering it may falsely trigger loss recoveries and reduce TCP throughput significantly. This patch improves that by increasing and reducing the reordering window based on DSACK, which is now supported in major TCP implementations. It makes RACK's reo_wnd adaptive based on DSACK and no. of recoveries. - If DSACK is received, increment reo_wnd by min_rtt/4 (upper bounded by srtt), since there is possibility that spurious retransmission was due to reordering delay longer than reo_wnd. - Persist the current reo_wnd value for TCP_RACK_RECOVERY_THRESH (16) no. of successful recoveries (accounts for full DSACK-based loss recovery undo). After that, reset it to default (min_rtt/4). - At max, reo_wnd is incremented only once per rtt. So that the new DSACK on which we are reacting, is due to the spurious retx (approx) after the reo_wnd has been updated last time. - reo_wnd is tracked in terms of steps (of min_rtt/4), rather than absolute value to account for change in rtt. In our internal testing, we observed significant increase in throughput, in scenarios where reordering exceeds min_rtt/4 (previous static value). Signed-off-by: Priyaranjan Jha <priyarjha@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-11-04 06:38:48 +07:00
u32 last_delivered; /* tp->delivered at last reo_wnd adj */
u8 reo_wnd_steps; /* Allowed reordering window */
#define TCP_RACK_RECOVERY_THRESH 16
u8 reo_wnd_persist:5, /* No. of recovery since last adj */
dsack_seen:1, /* Whether DSACK seen after last adj */
advanced:1, /* mstamp advanced since last lost marking */
reord:1; /* reordering detected */
tcp: track the packet timings in RACK This patch is the first half of the RACK loss recovery. RACK loss recovery uses the notion of time instead of packet sequence (FACK) or counts (dupthresh). It's inspired by the previous FACK heuristic in tcp_mark_lost_retrans(): when a limited transmit (new data packet) is sacked, then current retransmitted sequence below the newly sacked sequence must been lost, since at least one round trip time has elapsed. But it has several limitations: 1) can't detect tail drops since it depends on limited transmit 2) is disabled upon reordering (assumes no reordering) 3) only enabled in fast recovery ut not timeout recovery RACK (Recently ACK) addresses these limitations with the notion of time instead: a packet P1 is lost if a later packet P2 is s/acked, as at least one round trip has passed. Since RACK cares about the time sequence instead of the data sequence of packets, it can detect tail drops when later retransmission is s/acked while FACK or dupthresh can't. For reordering RACK uses a dynamically adjusted reordering window ("reo_wnd") to reduce false positives on ever (small) degree of reordering. This patch implements tcp_advanced_rack() which tracks the most recent transmission time among the packets that have been delivered (ACKed or SACKed) in tp->rack.mstamp. This timestamp is the key to determine which packet has been lost. Consider an example that the sender sends six packets: T1: P1 (lost) T2: P2 T3: P3 T4: P4 T100: sack of P2. rack.mstamp = T2 T101: retransmit P1 T102: sack of P2,P3,P4. rack.mstamp = T4 T205: ACK of P4 since the hole is repaired. rack.mstamp = T101 We need to be careful about spurious retransmission because it may falsely advance tp->rack.mstamp by an RTT or an RTO, causing RACK to falsely mark all packets lost, just like a spurious timeout. We identify spurious retransmission by the ACK's TS echo value. If TS option is not applicable but the retransmission is acknowledged less than min-RTT ago, it is likely to be spurious. We refrain from using the transmission time of these spurious retransmissions. The second half is implemented in the next patch that marks packet lost using RACK timestamp. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-10-17 11:57:46 +07:00
} rack;
tcp: Reorganize tcp_sock to fill 64-bit holes & improve locality I tried to group recovery related fields nearby (non-CA_Open related variables, to be more accurate) so that one to three cachelines would not be necessary in CA_Open. These are now contiguously deployed: struct sk_buff_head out_of_order_queue; /* 1968 80 */ /* --- cacheline 32 boundary (2048 bytes) --- */ struct tcp_sack_block duplicate_sack[1]; /* 2048 8 */ struct tcp_sack_block selective_acks[4]; /* 2056 32 */ struct tcp_sack_block recv_sack_cache[4]; /* 2088 32 */ /* --- cacheline 33 boundary (2112 bytes) was 8 bytes ago --- */ struct sk_buff * highest_sack; /* 2120 8 */ int lost_cnt_hint; /* 2128 4 */ int retransmit_cnt_hint; /* 2132 4 */ u32 lost_retrans_low; /* 2136 4 */ u8 reordering; /* 2140 1 */ u8 keepalive_probes; /* 2141 1 */ /* XXX 2 bytes hole, try to pack */ u32 prior_ssthresh; /* 2144 4 */ u32 high_seq; /* 2148 4 */ u32 retrans_stamp; /* 2152 4 */ u32 undo_marker; /* 2156 4 */ int undo_retrans; /* 2160 4 */ u32 total_retrans; /* 2164 4 */ ...and they're then followed by URG slowpath & keepalive related variables. Head of the out_of_order_queue always needed for empty checks, if that's empty (and TCP is in CA_Open), following ~200 bytes (in 64-bit) shouldn't be necessary for anything. If only OFO queue exists but TCP is in CA_Open, selective_acks (and possibly duplicate_sack) are necessary besides the out_of_order_queue but the rest of the block again shouldn't be (ie., the other direction had losses). As the cacheline boundaries depend on many factors in the preceeding stuff, trying to align considering them doesn't make too much sense. Commented one ordering hazard. There are number of low utilized u8/16s that could be combined get 2 bytes less in total so that the hole could be made to vanish (includes at least ecn_flags, urg_data, urg_mode, frto_counter, nonagle). Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Acked-by: Eric Dumazet <dada1@cosmosbay.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2008-05-29 17:25:23 +07:00
u16 advmss; /* Advertised MSS */
u8 compressed_ack;
u32 chrono_start; /* Start time in jiffies of a TCP chrono */
u32 chrono_stat[3]; /* Time in jiffies for chrono_stat stats */
u8 chrono_type:2, /* current chronograph type */
rate_app_limited:1, /* rate_{delivered,interval_us} limited? */
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 01:59:22 +07:00
fastopen_connect:1, /* FASTOPEN_CONNECT sockopt */
fastopen_no_cookie:1, /* Allow send/recv SYN+data without a cookie */
is_sack_reneg:1, /* in recovery from loss with SACK reneg? */
unused:2;
u8 nonagle : 4,/* Disable Nagle algorithm? */
thin_lto : 1,/* Use linear timeouts for thin streams */
tcp: send in-queue bytes in cmsg upon read Applications with many concurrent connections, high variance in receive queue length and tight memory bounds cannot allocate worst-case buffer size to drain sockets. Knowing the size of receive queue length, applications can optimize how they allocate buffers to read from the socket. The number of bytes pending on the socket is directly available through ioctl(FIONREAD/SIOCINQ) and can be approximated using getsockopt(MEMINFO) (rmem_alloc includes skb overheads in addition to application data). But, both of these options add an extra syscall per recvmsg. Moreover, ioctl(FIONREAD/SIOCINQ) takes the socket lock. Add the TCP_INQ socket option to TCP. When this socket option is set, recvmsg() relays the number of bytes available on the socket for reading to the application via the TCP_CM_INQ control message. Calculate the number of bytes after releasing the socket lock to include the processed backlog, if any. To avoid an extra branch in the hot path of recvmsg() for this new control message, move all cmsg processing inside an existing branch for processing receive timestamps. Since the socket lock is not held when calculating the size of receive queue, TCP_INQ is a hint. For example, it can overestimate the queue size by one byte, if FIN is received. With this method, applications can start reading from the socket using a small buffer, and then use larger buffers based on the remaining data when needed. V3 change-log: As suggested by David Miller, added loads with barrier to check whether we have multiple threads calling recvmsg in parallel. When that happens we lock the socket to calculate inq. V4 change-log: Removed inline from a static function. Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Willem de Bruijn <willemb@google.com> Reviewed-by: Eric Dumazet <edumazet@google.com> Reviewed-by: Neal Cardwell <ncardwell@google.com> Suggested-by: David Miller <davem@davemloft.net> Signed-off-by: David S. Miller <davem@davemloft.net>
2018-05-02 02:39:15 +07:00
recvmsg_inq : 1,/* Indicate # of bytes in queue upon recvmsg */
tcp: implement RFC5682 F-RTO This patch implements F-RTO (foward RTO recovery): When the first retransmission after timeout is acknowledged, F-RTO sends new data instead of old data. If the next ACK acknowledges some never-retransmitted data, then the timeout was spurious and the congestion state is reverted. Otherwise if the next ACK selectively acknowledges the new data, then the timeout was genuine and the loss recovery continues. This idea applies to recurring timeouts as well. While F-RTO sends different data during timeout recovery, it does not (and should not) change the congestion control. The implementaion follows the three steps of SACK enhanced algorithm (section 3) in RFC5682. Step 1 is in tcp_enter_loss(). Step 2 and 3 are in tcp_process_loss(). The basic version is not supported because SACK enhanced version also works for non-SACK connections. The new implementation is functionally in parity with the old F-RTO implementation except the one case where it increases undo events: In addition to the RFC algorithm, a spurious timeout may be detected without sending data in step 2, as long as the SACK confirms not all the original data are dropped. When this happens, the sender will undo the cwnd and perhaps enter fast recovery instead. This additional check increases the F-RTO undo events by 5x compared to the prior implementation on Google Web servers, since the sender often does not have new data to send for HTTP. Note F-RTO may detect spurious timeout before Eifel with timestamps does so. Signed-off-by: Yuchung Cheng <ycheng@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2013-03-20 20:33:00 +07:00
repair : 1,
frto : 1;/* F-RTO (RFC5682) activated in CA_Loss */
u8 repair_queue;
u8 syn_data:1, /* SYN includes data */
syn_fastopen:1, /* SYN includes Fast Open option */
syn_fastopen_exp:1,/* SYN includes Fast Open exp. option */
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 04:45:46 +07:00
syn_fastopen_ch:1, /* Active TFO re-enabling probe */
syn_data_acked:1,/* data in SYN is acked by SYN-ACK */
save_syn:1, /* Save headers of SYN packet */
is_cwnd_limited:1,/* forward progress limited by snd_cwnd? */
syn_smc:1; /* SYN includes SMC */
u32 tlp_high_seq; /* snd_nxt at the time of TLP retransmit. */
/* RTT measurement */
u64 tcp_mstamp; /* most recent packet received/sent */
tcp: switch rtt estimations to usec resolution Upcoming congestion controls for TCP require usec resolution for RTT estimations. Millisecond resolution is simply not enough these days. FQ/pacing in DC environments also require this change for finer control and removal of bimodal behavior due to the current hack in tcp_update_pacing_rate() for 'small rtt' TCP_CONG_RTT_STAMP is no longer needed. As Julian Anastasov pointed out, we need to keep user compatibility : tcp_metrics used to export RTT and RTTVAR in msec resolution, so we added RTT_US and RTTVAR_US. An iproute2 patch is needed to use the new attributes if provided by the kernel. In this example ss command displays a srtt of 32 usecs (10Gbit link) lpk51:~# ./ss -i dst lpk52 Netid State Recv-Q Send-Q Local Address:Port Peer Address:Port tcp ESTAB 0 1 10.246.11.51:42959 10.246.11.52:64614 cubic wscale:6,6 rto:201 rtt:0.032/0.001 ato:40 mss:1448 cwnd:10 send 3620.0Mbps pacing_rate 7240.0Mbps unacked:1 rcv_rtt:993 rcv_space:29559 Updated iproute2 ip command displays : lpk51:~# ./ip tcp_metrics | grep 10.246.11.52 10.246.11.52 age 561.914sec cwnd 10 rtt 274us rttvar 213us source 10.246.11.51 Old binary displays : lpk51:~# ip tcp_metrics | grep 10.246.11.52 10.246.11.52 age 561.914sec cwnd 10 rtt 250us rttvar 125us source 10.246.11.51 With help from Julian Anastasov, Stephen Hemminger and Yuchung Cheng Signed-off-by: Eric Dumazet <edumazet@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Cc: Stephen Hemminger <stephen@networkplumber.org> Cc: Yuchung Cheng <ycheng@google.com> Cc: Larry Brakmo <brakmo@google.com> Cc: Julian Anastasov <ja@ssi.bg> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-02-27 05:02:48 +07:00
u32 srtt_us; /* smoothed round trip time << 3 in usecs */
u32 mdev_us; /* medium deviation */
u32 mdev_max_us; /* maximal mdev for the last rtt period */
u32 rttvar_us; /* smoothed mdev_max */
u32 rtt_seq; /* sequence number to update rttvar */
struct minmax rtt_min;
u32 packets_out; /* Packets which are "in flight" */
u32 retrans_out; /* Retransmitted packets out */
u32 max_packets_out; /* max packets_out in last window */
u32 max_packets_seq; /* right edge of max_packets_out flight */
tcp: Reorganize tcp_sock to fill 64-bit holes & improve locality I tried to group recovery related fields nearby (non-CA_Open related variables, to be more accurate) so that one to three cachelines would not be necessary in CA_Open. These are now contiguously deployed: struct sk_buff_head out_of_order_queue; /* 1968 80 */ /* --- cacheline 32 boundary (2048 bytes) --- */ struct tcp_sack_block duplicate_sack[1]; /* 2048 8 */ struct tcp_sack_block selective_acks[4]; /* 2056 32 */ struct tcp_sack_block recv_sack_cache[4]; /* 2088 32 */ /* --- cacheline 33 boundary (2112 bytes) was 8 bytes ago --- */ struct sk_buff * highest_sack; /* 2120 8 */ int lost_cnt_hint; /* 2128 4 */ int retransmit_cnt_hint; /* 2132 4 */ u32 lost_retrans_low; /* 2136 4 */ u8 reordering; /* 2140 1 */ u8 keepalive_probes; /* 2141 1 */ /* XXX 2 bytes hole, try to pack */ u32 prior_ssthresh; /* 2144 4 */ u32 high_seq; /* 2148 4 */ u32 retrans_stamp; /* 2152 4 */ u32 undo_marker; /* 2156 4 */ int undo_retrans; /* 2160 4 */ u32 total_retrans; /* 2164 4 */ ...and they're then followed by URG slowpath & keepalive related variables. Head of the out_of_order_queue always needed for empty checks, if that's empty (and TCP is in CA_Open), following ~200 bytes (in 64-bit) shouldn't be necessary for anything. If only OFO queue exists but TCP is in CA_Open, selective_acks (and possibly duplicate_sack) are necessary besides the out_of_order_queue but the rest of the block again shouldn't be (ie., the other direction had losses). As the cacheline boundaries depend on many factors in the preceeding stuff, trying to align considering them doesn't make too much sense. Commented one ordering hazard. There are number of low utilized u8/16s that could be combined get 2 bytes less in total so that the hole could be made to vanish (includes at least ecn_flags, urg_data, urg_mode, frto_counter, nonagle). Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Acked-by: Eric Dumazet <dada1@cosmosbay.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2008-05-29 17:25:23 +07:00
u16 urg_data; /* Saved octet of OOB data and control flags */
u8 ecn_flags; /* ECN status bits. */
u8 keepalive_probes; /* num of allowed keep alive probes */
u32 reordering; /* Packet reordering metric. */
u32 snd_up; /* Urgent pointer */
/*
* Options received (usually on last packet, some only on SYN packets).
*/
struct tcp_options_received rx_opt;
/*
* Slow start and congestion control (see also Nagle, and Karn & Partridge)
*/
u32 snd_ssthresh; /* Slow start size threshold */
u32 snd_cwnd; /* Sending congestion window */
u32 snd_cwnd_cnt; /* Linear increase counter */
u32 snd_cwnd_clamp; /* Do not allow snd_cwnd to grow above this */
u32 snd_cwnd_used;
u32 snd_cwnd_stamp;
u32 prior_cwnd; /* cwnd right before starting loss recovery */
Proportional Rate Reduction for TCP. This patch implements Proportional Rate Reduction (PRR) for TCP. PRR is an algorithm that determines TCP's sending rate in fast recovery. PRR avoids excessive window reductions and aims for the actual congestion window size at the end of recovery to be as close as possible to the window determined by the congestion control algorithm. PRR also improves accuracy of the amount of data sent during loss recovery. The patch implements the recommended flavor of PRR called PRR-SSRB (Proportional rate reduction with slow start reduction bound) and replaces the existing rate halving algorithm. PRR improves upon the existing Linux fast recovery under a number of conditions including: 1) burst losses where the losses implicitly reduce the amount of outstanding data (pipe) below the ssthresh value selected by the congestion control algorithm and, 2) losses near the end of short flows where application runs out of data to send. As an example, with the existing rate halving implementation a single loss event can cause a connection carrying short Web transactions to go into the slow start mode after the recovery. This is because during recovery Linux pulls the congestion window down to packets_in_flight+1 on every ACK. A short Web response often runs out of new data to send and its pipe reduces to zero by the end of recovery when all its packets are drained from the network. Subsequent HTTP responses using the same connection will have to slow start to raise cwnd to ssthresh. PRR on the other hand aims for the cwnd to be as close as possible to ssthresh by the end of recovery. A description of PRR and a discussion of its performance can be found at the following links: - IETF Draft: http://tools.ietf.org/html/draft-mathis-tcpm-proportional-rate-reduction-01 - IETF Slides: http://www.ietf.org/proceedings/80/slides/tcpm-6.pdf http://tools.ietf.org/agenda/81/slides/tcpm-2.pdf - Paper to appear in Internet Measurements Conference (IMC) 2011: Improving TCP Loss Recovery Nandita Dukkipati, Matt Mathis, Yuchung Cheng Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2011-08-22 03:21:57 +07:00
u32 prr_delivered; /* Number of newly delivered packets to
* receiver in Recovery. */
u32 prr_out; /* Total number of pkts sent during Recovery. */
tcp: new delivery accounting This patch changes the accounting of how many packets are newly acked or sacked when the sender receives an ACK. The current approach basically computes newly_acked_sacked = (prior_packets - prior_sacked) - (tp->packets_out - tp->sacked_out) where prior_packets and prior_sacked out are snapshot at the beginning of the ACK processing. The new approach tracks the delivery information via a new TCP state variable "delivered" which monotically increases as new packets are delivered in order or out-of-order. The reason for this change is that the current approach is brittle that produces negative or inaccurate estimate. 1) For non-SACK connections, an ACK that advances the SND.UNA could reset the DUPACK counters (tp->sacked_out) in tcp_process_loss() or tcp_fastretrans_alert(). This inflates the inflight suddenly and causes under-estimate or even negative estimate. Here is a real example: before after (processing ACK) packets_out 75 73 sacked_out 23 0 ca state Loss Open The old approach computes (75-23) - (73 - 0) = -21 delivered while the new approach computes 1 delivered since it considers the 2nd-24th packets are delivered OOO. 2) MSS change would re-count packets_out and sacked_out so the estimate is in-accurate and can even become negative. E.g., the inflight is doubled when MSS is halved. 3) Spurious retransmission signaled by DSACK is not accounted The new approach is simpler and more robust. For SACK connections, tp->delivered increments as packets are being acked or sacked in SACK and ACK processing. For non-sack connections, it's done in tcp_remove_reno_sacks() and tcp_add_reno_sack(). When an ACK advances the SND.UNA, tp->delivered is incremented by the number of packets ACKed (less the current number of DUPACKs received plus one packet hole). Upon receiving a DUPACK, tp->delivered is incremented assuming one out-of-order packet is delivered. Upon receiving a DSACK, tp->delivered is incremtened assuming one retransmission is delivered in tcp_sacktag_write_queue(). Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Eric Dumazet <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-02-03 01:33:06 +07:00
u32 delivered; /* Total data packets delivered incl. rexmits */
u32 delivered_ce; /* Like the above but only ECE marked packets */
u32 lost; /* Total data packets lost incl. rexmits */
tcp: track application-limited rate samples This commit adds code to track whether the delivery rate represented by each rate_sample was limited by the application. Upon each transmit, we store in the is_app_limited field in the skb a boolean bit indicating whether there is a known "bubble in the pipe": a point in the rate sample interval where the sender was application-limited, and did not transmit even though the cwnd and pacing rate allowed it. This logic marks the flow app-limited on a write if *all* of the following are true: 1) There is less than 1 MSS of unsent data in the write queue available to transmit. 2) There is no packet in the sender's queues (e.g. in fq or the NIC tx queue). 3) The connection is not limited by cwnd. 4) There are no lost packets to retransmit. The tcp_rate_check_app_limited() code in tcp_rate.c determines whether the connection is application-limited at the moment. If the flow is application-limited, it sets the tp->app_limited field. If the flow is application-limited then that means there is effectively a "bubble" of silence in the pipe now, and this silence will be reflected in a lower bandwidth sample for any rate samples from now until we get an ACK indicating this bubble has exited the pipe: specifically, until we get an ACK for the next packet we transmit. When we send every skb we record in scb->tx.is_app_limited whether the resulting rate sample will be application-limited. The code in tcp_rate_gen() checks to see when it is safe to mark all known application-limited bubbles of silence as having exited the pipe. It does this by checking to see when the delivered count moves past the tp->app_limited marker. At this point it zeroes the tp->app_limited marker, as all known bubbles are out of the pipe. We make room for the tx.is_app_limited bit in the skb by borrowing a bit from the in_flight field used by NV to record the number of bytes in flight. The receive window in the TCP header is 16 bits, and the max receive window scaling shift factor is 14 (RFC 1323). So the max receive window offered by the TCP protocol is 2^(16+14) = 2^30. So we only need 30 bits for the tx.in_flight used by NV. Signed-off-by: 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 10:39:15 +07:00
u32 app_limited; /* limited until "delivered" reaches this val */
u64 first_tx_mstamp; /* start of window send phase */
u64 delivered_mstamp; /* time we reached "delivered" */
u32 rate_delivered; /* saved rate sample: packets delivered */
u32 rate_interval_us; /* saved rate sample: time elapsed */
u32 rcv_wnd; /* Current receiver window */
u32 write_seq; /* Tail(+1) of data held in tcp send buffer */
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 10:27:07 +07:00
u32 notsent_lowat; /* TCP_NOTSENT_LOWAT */
u32 pushed_seq; /* Last pushed seq, required to talk to windows */
tcp: Reorganize tcp_sock to fill 64-bit holes & improve locality I tried to group recovery related fields nearby (non-CA_Open related variables, to be more accurate) so that one to three cachelines would not be necessary in CA_Open. These are now contiguously deployed: struct sk_buff_head out_of_order_queue; /* 1968 80 */ /* --- cacheline 32 boundary (2048 bytes) --- */ struct tcp_sack_block duplicate_sack[1]; /* 2048 8 */ struct tcp_sack_block selective_acks[4]; /* 2056 32 */ struct tcp_sack_block recv_sack_cache[4]; /* 2088 32 */ /* --- cacheline 33 boundary (2112 bytes) was 8 bytes ago --- */ struct sk_buff * highest_sack; /* 2120 8 */ int lost_cnt_hint; /* 2128 4 */ int retransmit_cnt_hint; /* 2132 4 */ u32 lost_retrans_low; /* 2136 4 */ u8 reordering; /* 2140 1 */ u8 keepalive_probes; /* 2141 1 */ /* XXX 2 bytes hole, try to pack */ u32 prior_ssthresh; /* 2144 4 */ u32 high_seq; /* 2148 4 */ u32 retrans_stamp; /* 2152 4 */ u32 undo_marker; /* 2156 4 */ int undo_retrans; /* 2160 4 */ u32 total_retrans; /* 2164 4 */ ...and they're then followed by URG slowpath & keepalive related variables. Head of the out_of_order_queue always needed for empty checks, if that's empty (and TCP is in CA_Open), following ~200 bytes (in 64-bit) shouldn't be necessary for anything. If only OFO queue exists but TCP is in CA_Open, selective_acks (and possibly duplicate_sack) are necessary besides the out_of_order_queue but the rest of the block again shouldn't be (ie., the other direction had losses). As the cacheline boundaries depend on many factors in the preceeding stuff, trying to align considering them doesn't make too much sense. Commented one ordering hazard. There are number of low utilized u8/16s that could be combined get 2 bytes less in total so that the hole could be made to vanish (includes at least ecn_flags, urg_data, urg_mode, frto_counter, nonagle). Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Acked-by: Eric Dumazet <dada1@cosmosbay.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2008-05-29 17:25:23 +07:00
u32 lost_out; /* Lost packets */
u32 sacked_out; /* SACK'd packets */
tcp: internal implementation for pacing BBR congestion control depends on pacing, and pacing is currently handled by sch_fq packet scheduler for performance reasons, and also because implemening pacing with FQ was convenient to truly avoid bursts. However there are many cases where this packet scheduler constraint is not practical. - Many linux hosts are not focusing on handling thousands of TCP flows in the most efficient way. - Some routers use fq_codel or other AQM, but still would like to use BBR for the few TCP flows they initiate/terminate. This patch implements an automatic fallback to internal pacing. Pacing is requested either by BBR or use of SO_MAX_PACING_RATE option. If sch_fq happens to be in the egress path, pacing is delegated to the qdisc, otherwise pacing is done by TCP itself. One advantage of pacing from TCP stack is to get more precise rtt estimations, and less work done from TX completion, since TCP Small queue limits are not generally hit. Setups with single TX queue but many cpus might even benefit from this. Note that unlike sch_fq, we do not take into account header sizes. Taking care of these headers would add additional complexity for no practical differences in behavior. Some performance numbers using 800 TCP_STREAM flows rate limited to ~48 Mbit per second on 40Gbit NIC. If MQ+pfifo_fast is used on the NIC : $ sar -n DEV 1 5 | grep eth 14:48:44 eth0 725743.00 2932134.00 46776.76 4335184.68 0.00 0.00 1.00 14:48:45 eth0 725349.00 2932112.00 46751.86 4335158.90 0.00 0.00 0.00 14:48:46 eth0 725101.00 2931153.00 46735.07 4333748.63 0.00 0.00 0.00 14:48:47 eth0 725099.00 2931161.00 46735.11 4333760.44 0.00 0.00 1.00 14:48:48 eth0 725160.00 2931731.00 46738.88 4334606.07 0.00 0.00 0.00 Average: eth0 725290.40 2931658.20 46747.54 4334491.74 0.00 0.00 0.40 $ vmstat 1 5 procs -----------memory---------- ---swap-- -----io---- -system-- ------cpu----- r b swpd free buff cache si so bi bo in cs us sy id wa st 4 0 0 259825920 45644 2708324 0 0 21 2 247 98 0 0 100 0 0 4 0 0 259823744 45644 2708356 0 0 0 0 2400825 159843 0 19 81 0 0 0 0 0 259824208 45644 2708072 0 0 0 0 2407351 159929 0 19 81 0 0 1 0 0 259824592 45644 2708128 0 0 0 0 2405183 160386 0 19 80 0 0 1 0 0 259824272 45644 2707868 0 0 0 32 2396361 158037 0 19 81 0 0 Now use MQ+FQ : lpaa23:~# echo fq >/proc/sys/net/core/default_qdisc lpaa23:~# tc qdisc replace dev eth0 root mq $ sar -n DEV 1 5 | grep eth 14:49:57 eth0 678614.00 2727930.00 43739.13 4033279.14 0.00 0.00 0.00 14:49:58 eth0 677620.00 2723971.00 43674.69 4027429.62 0.00 0.00 1.00 14:49:59 eth0 676396.00 2719050.00 43596.83 4020125.02 0.00 0.00 0.00 14:50:00 eth0 675197.00 2714173.00 43518.62 4012938.90 0.00 0.00 1.00 14:50:01 eth0 676388.00 2719063.00 43595.47 4020171.64 0.00 0.00 0.00 Average: eth0 676843.00 2720837.40 43624.95 4022788.86 0.00 0.00 0.40 $ vmstat 1 5 procs -----------memory---------- ---swap-- -----io---- -system-- ------cpu----- r b swpd free buff cache si so bi bo in cs us sy id wa st 2 0 0 259832240 46008 2710912 0 0 21 2 223 192 0 1 99 0 0 1 0 0 259832896 46008 2710744 0 0 0 0 1702206 198078 0 17 82 0 0 0 0 0 259830272 46008 2710596 0 0 0 0 1696340 197756 1 17 83 0 0 4 0 0 259829168 46024 2710584 0 0 16 0 1688472 197158 1 17 82 0 0 3 0 0 259830224 46024 2710408 0 0 0 0 1692450 197212 0 18 82 0 0 As expected, number of interrupts per second is very different. Signed-off-by: Eric Dumazet <edumazet@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Van Jacobson <vanj@google.com> Cc: Jerry Chu <hkchu@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-05-16 18:24:36 +07:00
struct hrtimer pacing_timer;
struct hrtimer compressed_ack_timer;
tcp: internal implementation for pacing BBR congestion control depends on pacing, and pacing is currently handled by sch_fq packet scheduler for performance reasons, and also because implemening pacing with FQ was convenient to truly avoid bursts. However there are many cases where this packet scheduler constraint is not practical. - Many linux hosts are not focusing on handling thousands of TCP flows in the most efficient way. - Some routers use fq_codel or other AQM, but still would like to use BBR for the few TCP flows they initiate/terminate. This patch implements an automatic fallback to internal pacing. Pacing is requested either by BBR or use of SO_MAX_PACING_RATE option. If sch_fq happens to be in the egress path, pacing is delegated to the qdisc, otherwise pacing is done by TCP itself. One advantage of pacing from TCP stack is to get more precise rtt estimations, and less work done from TX completion, since TCP Small queue limits are not generally hit. Setups with single TX queue but many cpus might even benefit from this. Note that unlike sch_fq, we do not take into account header sizes. Taking care of these headers would add additional complexity for no practical differences in behavior. Some performance numbers using 800 TCP_STREAM flows rate limited to ~48 Mbit per second on 40Gbit NIC. If MQ+pfifo_fast is used on the NIC : $ sar -n DEV 1 5 | grep eth 14:48:44 eth0 725743.00 2932134.00 46776.76 4335184.68 0.00 0.00 1.00 14:48:45 eth0 725349.00 2932112.00 46751.86 4335158.90 0.00 0.00 0.00 14:48:46 eth0 725101.00 2931153.00 46735.07 4333748.63 0.00 0.00 0.00 14:48:47 eth0 725099.00 2931161.00 46735.11 4333760.44 0.00 0.00 1.00 14:48:48 eth0 725160.00 2931731.00 46738.88 4334606.07 0.00 0.00 0.00 Average: eth0 725290.40 2931658.20 46747.54 4334491.74 0.00 0.00 0.40 $ vmstat 1 5 procs -----------memory---------- ---swap-- -----io---- -system-- ------cpu----- r b swpd free buff cache si so bi bo in cs us sy id wa st 4 0 0 259825920 45644 2708324 0 0 21 2 247 98 0 0 100 0 0 4 0 0 259823744 45644 2708356 0 0 0 0 2400825 159843 0 19 81 0 0 0 0 0 259824208 45644 2708072 0 0 0 0 2407351 159929 0 19 81 0 0 1 0 0 259824592 45644 2708128 0 0 0 0 2405183 160386 0 19 80 0 0 1 0 0 259824272 45644 2707868 0 0 0 32 2396361 158037 0 19 81 0 0 Now use MQ+FQ : lpaa23:~# echo fq >/proc/sys/net/core/default_qdisc lpaa23:~# tc qdisc replace dev eth0 root mq $ sar -n DEV 1 5 | grep eth 14:49:57 eth0 678614.00 2727930.00 43739.13 4033279.14 0.00 0.00 0.00 14:49:58 eth0 677620.00 2723971.00 43674.69 4027429.62 0.00 0.00 1.00 14:49:59 eth0 676396.00 2719050.00 43596.83 4020125.02 0.00 0.00 0.00 14:50:00 eth0 675197.00 2714173.00 43518.62 4012938.90 0.00 0.00 1.00 14:50:01 eth0 676388.00 2719063.00 43595.47 4020171.64 0.00 0.00 0.00 Average: eth0 676843.00 2720837.40 43624.95 4022788.86 0.00 0.00 0.40 $ vmstat 1 5 procs -----------memory---------- ---swap-- -----io---- -system-- ------cpu----- r b swpd free buff cache si so bi bo in cs us sy id wa st 2 0 0 259832240 46008 2710912 0 0 21 2 223 192 0 1 99 0 0 1 0 0 259832896 46008 2710744 0 0 0 0 1702206 198078 0 17 82 0 0 0 0 0 259830272 46008 2710596 0 0 0 0 1696340 197756 1 17 83 0 0 4 0 0 259829168 46024 2710584 0 0 16 0 1688472 197158 1 17 82 0 0 3 0 0 259830224 46024 2710408 0 0 0 0 1692450 197212 0 18 82 0 0 As expected, number of interrupts per second is very different. Signed-off-by: Eric Dumazet <edumazet@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Van Jacobson <vanj@google.com> Cc: Jerry Chu <hkchu@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-05-16 18:24:36 +07:00
tcp: Reorganize tcp_sock to fill 64-bit holes & improve locality I tried to group recovery related fields nearby (non-CA_Open related variables, to be more accurate) so that one to three cachelines would not be necessary in CA_Open. These are now contiguously deployed: struct sk_buff_head out_of_order_queue; /* 1968 80 */ /* --- cacheline 32 boundary (2048 bytes) --- */ struct tcp_sack_block duplicate_sack[1]; /* 2048 8 */ struct tcp_sack_block selective_acks[4]; /* 2056 32 */ struct tcp_sack_block recv_sack_cache[4]; /* 2088 32 */ /* --- cacheline 33 boundary (2112 bytes) was 8 bytes ago --- */ struct sk_buff * highest_sack; /* 2120 8 */ int lost_cnt_hint; /* 2128 4 */ int retransmit_cnt_hint; /* 2132 4 */ u32 lost_retrans_low; /* 2136 4 */ u8 reordering; /* 2140 1 */ u8 keepalive_probes; /* 2141 1 */ /* XXX 2 bytes hole, try to pack */ u32 prior_ssthresh; /* 2144 4 */ u32 high_seq; /* 2148 4 */ u32 retrans_stamp; /* 2152 4 */ u32 undo_marker; /* 2156 4 */ int undo_retrans; /* 2160 4 */ u32 total_retrans; /* 2164 4 */ ...and they're then followed by URG slowpath & keepalive related variables. Head of the out_of_order_queue always needed for empty checks, if that's empty (and TCP is in CA_Open), following ~200 bytes (in 64-bit) shouldn't be necessary for anything. If only OFO queue exists but TCP is in CA_Open, selective_acks (and possibly duplicate_sack) are necessary besides the out_of_order_queue but the rest of the block again shouldn't be (ie., the other direction had losses). As the cacheline boundaries depend on many factors in the preceeding stuff, trying to align considering them doesn't make too much sense. Commented one ordering hazard. There are number of low utilized u8/16s that could be combined get 2 bytes less in total so that the hole could be made to vanish (includes at least ecn_flags, urg_data, urg_mode, frto_counter, nonagle). Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Acked-by: Eric Dumazet <dada1@cosmosbay.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2008-05-29 17:25:23 +07:00
/* from STCP, retrans queue hinting */
struct sk_buff* lost_skb_hint;
struct sk_buff *retransmit_skb_hint;
2016-09-08 04:49:28 +07:00
/* OOO segments go in this rbtree. Socket lock must be held. */
struct rb_root out_of_order_queue;
struct sk_buff *ooo_last_skb; /* cache rb_last(out_of_order_queue) */
tcp: Reorganize tcp_sock to fill 64-bit holes & improve locality I tried to group recovery related fields nearby (non-CA_Open related variables, to be more accurate) so that one to three cachelines would not be necessary in CA_Open. These are now contiguously deployed: struct sk_buff_head out_of_order_queue; /* 1968 80 */ /* --- cacheline 32 boundary (2048 bytes) --- */ struct tcp_sack_block duplicate_sack[1]; /* 2048 8 */ struct tcp_sack_block selective_acks[4]; /* 2056 32 */ struct tcp_sack_block recv_sack_cache[4]; /* 2088 32 */ /* --- cacheline 33 boundary (2112 bytes) was 8 bytes ago --- */ struct sk_buff * highest_sack; /* 2120 8 */ int lost_cnt_hint; /* 2128 4 */ int retransmit_cnt_hint; /* 2132 4 */ u32 lost_retrans_low; /* 2136 4 */ u8 reordering; /* 2140 1 */ u8 keepalive_probes; /* 2141 1 */ /* XXX 2 bytes hole, try to pack */ u32 prior_ssthresh; /* 2144 4 */ u32 high_seq; /* 2148 4 */ u32 retrans_stamp; /* 2152 4 */ u32 undo_marker; /* 2156 4 */ int undo_retrans; /* 2160 4 */ u32 total_retrans; /* 2164 4 */ ...and they're then followed by URG slowpath & keepalive related variables. Head of the out_of_order_queue always needed for empty checks, if that's empty (and TCP is in CA_Open), following ~200 bytes (in 64-bit) shouldn't be necessary for anything. If only OFO queue exists but TCP is in CA_Open, selective_acks (and possibly duplicate_sack) are necessary besides the out_of_order_queue but the rest of the block again shouldn't be (ie., the other direction had losses). As the cacheline boundaries depend on many factors in the preceeding stuff, trying to align considering them doesn't make too much sense. Commented one ordering hazard. There are number of low utilized u8/16s that could be combined get 2 bytes less in total so that the hole could be made to vanish (includes at least ecn_flags, urg_data, urg_mode, frto_counter, nonagle). Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Acked-by: Eric Dumazet <dada1@cosmosbay.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2008-05-29 17:25:23 +07:00
/* SACKs data, these 2 need to be together (see tcp_options_write) */
struct tcp_sack_block duplicate_sack[1]; /* D-SACK block */
struct tcp_sack_block selective_acks[4]; /* The SACKS themselves*/
struct tcp_sack_block recv_sack_cache[4];
struct sk_buff *highest_sack; /* skb just after the highest
* skb with SACKed bit set
* (validity guaranteed only if
* sacked_out > 0)
*/
int lost_cnt_hint;
u32 prior_ssthresh; /* ssthresh saved at recovery start */
u32 high_seq; /* snd_nxt at onset of congestion */
u32 retrans_stamp; /* Timestamp of the last retransmit,
* also used in SYN-SENT to remember stamp of
* the first SYN. */
u32 undo_marker; /* snd_una upon a new recovery episode. */
int undo_retrans; /* number of undoable retransmissions. */
tcp: Reorganize tcp_sock to fill 64-bit holes & improve locality I tried to group recovery related fields nearby (non-CA_Open related variables, to be more accurate) so that one to three cachelines would not be necessary in CA_Open. These are now contiguously deployed: struct sk_buff_head out_of_order_queue; /* 1968 80 */ /* --- cacheline 32 boundary (2048 bytes) --- */ struct tcp_sack_block duplicate_sack[1]; /* 2048 8 */ struct tcp_sack_block selective_acks[4]; /* 2056 32 */ struct tcp_sack_block recv_sack_cache[4]; /* 2088 32 */ /* --- cacheline 33 boundary (2112 bytes) was 8 bytes ago --- */ struct sk_buff * highest_sack; /* 2120 8 */ int lost_cnt_hint; /* 2128 4 */ int retransmit_cnt_hint; /* 2132 4 */ u32 lost_retrans_low; /* 2136 4 */ u8 reordering; /* 2140 1 */ u8 keepalive_probes; /* 2141 1 */ /* XXX 2 bytes hole, try to pack */ u32 prior_ssthresh; /* 2144 4 */ u32 high_seq; /* 2148 4 */ u32 retrans_stamp; /* 2152 4 */ u32 undo_marker; /* 2156 4 */ int undo_retrans; /* 2160 4 */ u32 total_retrans; /* 2164 4 */ ...and they're then followed by URG slowpath & keepalive related variables. Head of the out_of_order_queue always needed for empty checks, if that's empty (and TCP is in CA_Open), following ~200 bytes (in 64-bit) shouldn't be necessary for anything. If only OFO queue exists but TCP is in CA_Open, selective_acks (and possibly duplicate_sack) are necessary besides the out_of_order_queue but the rest of the block again shouldn't be (ie., the other direction had losses). As the cacheline boundaries depend on many factors in the preceeding stuff, trying to align considering them doesn't make too much sense. Commented one ordering hazard. There are number of low utilized u8/16s that could be combined get 2 bytes less in total so that the hole could be made to vanish (includes at least ecn_flags, urg_data, urg_mode, frto_counter, nonagle). Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Acked-by: Eric Dumazet <dada1@cosmosbay.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2008-05-29 17:25:23 +07:00
u32 total_retrans; /* Total retransmits for entire connection */
u32 urg_seq; /* Seq of received urgent pointer */
unsigned int keepalive_time; /* time before keep alive takes place */
unsigned int keepalive_intvl; /* time interval between keep alive probes */
int linger2;
/* Sock_ops bpf program related variables */
#ifdef CONFIG_BPF
u8 bpf_sock_ops_cb_flags; /* Control calling BPF programs
* values defined in uapi/linux/tcp.h
*/
#define BPF_SOCK_OPS_TEST_FLAG(TP, ARG) (TP->bpf_sock_ops_cb_flags & ARG)
#else
#define BPF_SOCK_OPS_TEST_FLAG(TP, ARG) 0
#endif
/* Receiver side RTT estimation */
struct {
u32 rtt_us;
u32 seq;
u64 time;
} rcv_rtt_est;
/* Receiver queue space */
struct {
u32 space;
u32 seq;
u64 time;
} rcvq_space;
/* TCP-specific MTU probe information. */
struct {
u32 probe_seq_start;
u32 probe_seq_end;
} mtu_probe;
u32 mtu_info; /* We received an ICMP_FRAG_NEEDED / ICMPV6_PKT_TOOBIG
* while socket was owned by user.
*/
#ifdef CONFIG_TCP_MD5SIG
/* TCP AF-Specific parts; only used by MD5 Signature support so far */
const struct tcp_sock_af_ops *af_specific;
/* TCP MD5 Signature Option information */
struct tcp_md5sig_info __rcu *md5sig_info;
#endif
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 01:17:05 +07:00
/* TCP fastopen related information */
struct tcp_fastopen_request *fastopen_req;
/* fastopen_rsk points to request_sock that resulted in this big
* socket. Used to retransmit SYNACKs etc.
*/
struct request_sock *fastopen_rsk;
u32 *saved_syn;
};
enum tsq_enum {
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 12:50:31 +07:00
TSQ_THROTTLED,
TSQ_QUEUED,
TCP_TSQ_DEFERRED, /* tcp_tasklet_func() found socket was owned */
TCP_WRITE_TIMER_DEFERRED, /* tcp_write_timer() found socket was owned */
TCP_DELACK_TIMER_DEFERRED, /* tcp_delack_timer() found socket was owned */
TCP_MTU_REDUCED_DEFERRED, /* tcp_v{4|6}_err() could not call
* tcp_v{4|6}_mtu_reduced()
*/
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 12:50:31 +07:00
};
enum tsq_flags {
TSQF_THROTTLED = (1UL << TSQ_THROTTLED),
TSQF_QUEUED = (1UL << TSQ_QUEUED),
TCPF_TSQ_DEFERRED = (1UL << TCP_TSQ_DEFERRED),
TCPF_WRITE_TIMER_DEFERRED = (1UL << TCP_WRITE_TIMER_DEFERRED),
TCPF_DELACK_TIMER_DEFERRED = (1UL << TCP_DELACK_TIMER_DEFERRED),
TCPF_MTU_REDUCED_DEFERRED = (1UL << TCP_MTU_REDUCED_DEFERRED),
};
static inline struct tcp_sock *tcp_sk(const struct sock *sk)
{
return (struct tcp_sock *)sk;
}
struct tcp_timewait_sock {
struct inet_timewait_sock tw_sk;
#define tw_rcv_nxt tw_sk.__tw_common.skc_tw_rcv_nxt
#define tw_snd_nxt tw_sk.__tw_common.skc_tw_snd_nxt
u32 tw_rcv_wnd;
u32 tw_ts_offset;
u32 tw_ts_recent;
/* The time we sent the last out-of-window ACK: */
u32 tw_last_oow_ack_time;
long tw_ts_recent_stamp;
#ifdef CONFIG_TCP_MD5SIG
struct tcp_md5sig_key *tw_md5_key;
#endif
};
static inline struct tcp_timewait_sock *tcp_twsk(const struct sock *sk)
{
return (struct tcp_timewait_sock *)sk;
}
static inline bool tcp_passive_fastopen(const struct sock *sk)
{
return (sk->sk_state == TCP_SYN_RECV &&
tcp_sk(sk)->fastopen_rsk != NULL);
}
static inline void fastopen_queue_tune(struct sock *sk, int backlog)
{
struct request_sock_queue *queue = &inet_csk(sk)->icsk_accept_queue;
int somaxconn = READ_ONCE(sock_net(sk)->core.sysctl_somaxconn);
queue->fastopenq.max_qlen = min_t(unsigned int, backlog, somaxconn);
}
static inline void tcp_move_syn(struct tcp_sock *tp,
struct request_sock *req)
{
tp->saved_syn = req->saved_syn;
req->saved_syn = NULL;
}
static inline void tcp_saved_syn_free(struct tcp_sock *tp)
{
kfree(tp->saved_syn);
tp->saved_syn = NULL;
}
struct sk_buff *tcp_get_timestamping_opt_stats(const struct sock *sk);
static inline u16 tcp_mss_clamp(const struct tcp_sock *tp, u16 mss)
{
/* We use READ_ONCE() here because socket might not be locked.
* This happens for listeners.
*/
u16 user_mss = READ_ONCE(tp->rx_opt.user_mss);
return (user_mss && user_mss < mss) ? user_mss : mss;
}
#endif /* _LINUX_TCP_H */