OpenCloudOS-Kernel/include/net/tcp.h

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/* SPDX-License-Identifier: GPL-2.0-or-later */
/*
* 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 module.
*
* Version: @(#)tcp.h 1.0.5 05/23/93
*
* Authors: Ross Biro
* Fred N. van Kempen, <waltje@uWalt.NL.Mugnet.ORG>
*/
#ifndef _TCP_H
#define _TCP_H
#define FASTRETRANS_DEBUG 1
#include <linux/list.h>
#include <linux/tcp.h>
#include <linux/bug.h>
#include <linux/slab.h>
#include <linux/cache.h>
#include <linux/percpu.h>
#include <linux/skbuff.h>
TCPCT part 1d: define TCP cookie option, extend existing struct's Data structures are carefully composed to require minimal additions. For example, the struct tcp_options_received cookie_plus variable fits between existing 16-bit and 8-bit variables, requiring no additional space (taking alignment into consideration). There are no additions to tcp_request_sock, and only 1 pointer in tcp_sock. This is a significantly revised implementation of an earlier (year-old) patch that no longer applies cleanly, with permission of the original author (Adam Langley): http://thread.gmane.org/gmane.linux.network/102586 The principle difference is using a TCP option to carry the cookie nonce, instead of a user configured offset in the data. This is more flexible and less subject to user configuration error. Such a cookie option has been suggested for many years, and is also useful without SYN data, allowing several related concepts to use the same extension option. "Re: SYN floods (was: does history repeat itself?)", September 9, 1996. http://www.merit.net/mail.archives/nanog/1996-09/msg00235.html "Re: what a new TCP header might look like", May 12, 1998. ftp://ftp.isi.edu/end2end/end2end-interest-1998.mail These functions will also be used in subsequent patches that implement additional features. Requires: TCPCT part 1a: add request_values parameter for sending SYNACK TCPCT part 1b: generate Responder Cookie secret TCPCT part 1c: sysctl_tcp_cookie_size, socket option TCP_COOKIE_TRANSACTIONS Signed-off-by: William.Allen.Simpson@gmail.com Signed-off-by: David S. Miller <davem@davemloft.net>
2009-12-03 02:17:05 +08:00
#include <linux/kref.h>
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 06:02:48 +08:00
#include <linux/ktime.h>
#include <linux/indirect_call_wrapper.h>
#include <net/inet_connection_sock.h>
#include <net/inet_timewait_sock.h>
#include <net/inet_hashtables.h>
#include <net/checksum.h>
#include <net/request_sock.h>
tcp: Avoid TCP syncookie rejected by SO_REUSEPORT socket Although the actual cookie check "__cookie_v[46]_check()" does not involve sk specific info, it checks whether the sk has recent synq overflow event in "tcp_synq_no_recent_overflow()". The tcp_sk(sk)->rx_opt.ts_recent_stamp is updated every second when it has sent out a syncookie (through "tcp_synq_overflow()"). The above per sk "recent synq overflow event timestamp" works well for non SO_REUSEPORT use case. However, it may cause random connection request reject/discard when SO_REUSEPORT is used with syncookie because it fails the "tcp_synq_no_recent_overflow()" test. When SO_REUSEPORT is used, it usually has multiple listening socks serving TCP connection requests destinated to the same local IP:PORT. There are cases that the TCP-ACK-COOKIE may not be received by the same sk that sent out the syncookie. For example, if reuse->socks[] began with {sk0, sk1}, 1) sk1 sent out syncookies and tcp_sk(sk1)->rx_opt.ts_recent_stamp was updated. 2) the reuse->socks[] became {sk1, sk2} later. e.g. sk0 was first closed and then sk2 was added. Here, sk2 does not have ts_recent_stamp set. There are other ordering that will trigger the similar situation below but the idea is the same. 3) When the TCP-ACK-COOKIE comes back, sk2 was selected. "tcp_synq_no_recent_overflow(sk2)" returns true. In this case, all syncookies sent by sk1 will be handled (and rejected) by sk2 while sk1 is still alive. The userspace may create and remove listening SO_REUSEPORT sockets as it sees fit. E.g. Adding new thread (and SO_REUSEPORT sock) to handle incoming requests, old process stopping and new process starting...etc. With or without SO_ATTACH_REUSEPORT_[CB]BPF, the sockets leaving and joining a reuseport group makes picking the same sk to check the syncookie very difficult (if not impossible). The later patches will allow bpf prog more flexibility in deciding where a sk should be located in a bpf map and selecting a particular SO_REUSEPORT sock as it sees fit. e.g. Without closing any sock, replace the whole bpf reuseport_array in one map_update() by using map-in-map. Getting the syncookie check working smoothly across socks in the same "reuse->socks[]" is important. A partial solution is to set the newly added sk's ts_recent_stamp to the max ts_recent_stamp of a reuseport group but that will require to iterate through reuse->socks[] OR pessimistically set it to "now - TCP_SYNCOOKIE_VALID" when a sk is joining a reuseport group. However, neither of them will solve the existing sk getting moved around the reuse->socks[] and that sk may not have ts_recent_stamp updated, unlikely under continuous synflood but not impossible. This patch opts to treat the reuseport group as a whole when considering the last synq overflow timestamp since they are serving the same IP:PORT from the userspace (and BPF program) perspective. "synq_overflow_ts" is added to "struct sock_reuseport". The tcp_synq_overflow() and tcp_synq_no_recent_overflow() will update/check reuse->synq_overflow_ts if the sk is in a reuseport group. Similar to the reuseport decision in __inet_lookup_listener(), both sk->sk_reuseport and sk->sk_reuseport_cb are tested for SO_REUSEPORT usage. Update on "synq_overflow_ts" happens at roughly once every second. A synflood test was done with a 16 rx-queues and 16 reuseport sockets. No meaningful performance change is observed. Before and after the change is ~9Mpps in IPv4. Cc: Eric Dumazet <edumazet@google.com> Signed-off-by: Martin KaFai Lau <kafai@fb.com> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
2018-08-08 16:01:21 +08:00
#include <net/sock_reuseport.h>
#include <net/sock.h>
#include <net/snmp.h>
#include <net/ip.h>
#include <net/tcp_states.h>
#include <net/inet_ecn.h>
#include <net/dst.h>
#include <net/mptcp.h>
#include <linux/seq_file.h>
#include <linux/memcontrol.h>
bpf: BPF support for sock_ops Created a new BPF program type, BPF_PROG_TYPE_SOCK_OPS, and a corresponding struct that allows BPF programs of this type to access some of the socket's fields (such as IP addresses, ports, etc.). It uses the existing bpf cgroups infrastructure so the programs can be attached per cgroup with full inheritance support. The program will be called at appropriate times to set relevant connections parameters such as buffer sizes, SYN and SYN-ACK RTOs, etc., based on connection information such as IP addresses, port numbers, etc. Alghough there are already 3 mechanisms to set parameters (sysctls, route metrics and setsockopts), this new mechanism provides some distinct advantages. Unlike sysctls, it can set parameters per connection. In contrast to route metrics, it can also use port numbers and information provided by a user level program. In addition, it could set parameters probabilistically for evaluation purposes (i.e. do something different on 10% of the flows and compare results with the other 90% of the flows). Also, in cases where IPv6 addresses contain geographic information, the rules to make changes based on the distance (or RTT) between the hosts are much easier than route metric rules and can be global. Finally, unlike setsockopt, it oes not require application changes and it can be updated easily at any time. Although the bpf cgroup framework already contains a sock related program type (BPF_PROG_TYPE_CGROUP_SOCK), I created the new type (BPF_PROG_TYPE_SOCK_OPS) beccause the existing type expects to be called only once during the connections's lifetime. In contrast, the new program type will be called multiple times from different places in the network stack code. For example, before sending SYN and SYN-ACKs to set an appropriate timeout, when the connection is established to set congestion control, etc. As a result it has "op" field to specify the type of operation requested. The purpose of this new program type is to simplify setting connection parameters, such as buffer sizes, TCP's SYN RTO, etc. For example, it is easy to use facebook's internal IPv6 addresses to determine if both hosts of a connection are in the same datacenter. Therefore, it is easy to write a BPF program to choose a small SYN RTO value when both hosts are in the same datacenter. This patch only contains the framework to support the new BPF program type, following patches add the functionality to set various connection parameters. This patch defines a new BPF program type: BPF_PROG_TYPE_SOCKET_OPS and a new bpf syscall command to load a new program of this type: BPF_PROG_LOAD_SOCKET_OPS. Two new corresponding structs (one for the kernel one for the user/BPF program): /* kernel version */ struct bpf_sock_ops_kern { struct sock *sk; __u32 op; union { __u32 reply; __u32 replylong[4]; }; }; /* user version * Some fields are in network byte order reflecting the sock struct * Use the bpf_ntohl helper macro in samples/bpf/bpf_endian.h to * convert them to host byte order. */ struct bpf_sock_ops { __u32 op; union { __u32 reply; __u32 replylong[4]; }; __u32 family; __u32 remote_ip4; /* In network byte order */ __u32 local_ip4; /* In network byte order */ __u32 remote_ip6[4]; /* In network byte order */ __u32 local_ip6[4]; /* In network byte order */ __u32 remote_port; /* In network byte order */ __u32 local_port; /* In host byte horder */ }; Currently there are two types of ops. The first type expects the BPF program to return a value which is then used by the caller (or a negative value to indicate the operation is not supported). The second type expects state changes to be done by the BPF program, for example through a setsockopt BPF helper function, and they ignore the return value. The reply fields of the bpf_sockt_ops struct are there in case a bpf program needs to return a value larger than an integer. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Daniel Borkmann <daniel@iogearbox.net> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-07-01 11:02:40 +08:00
#include <linux/bpf-cgroup.h>
#include <linux/siphash.h>
bpf: BPF support for sock_ops Created a new BPF program type, BPF_PROG_TYPE_SOCK_OPS, and a corresponding struct that allows BPF programs of this type to access some of the socket's fields (such as IP addresses, ports, etc.). It uses the existing bpf cgroups infrastructure so the programs can be attached per cgroup with full inheritance support. The program will be called at appropriate times to set relevant connections parameters such as buffer sizes, SYN and SYN-ACK RTOs, etc., based on connection information such as IP addresses, port numbers, etc. Alghough there are already 3 mechanisms to set parameters (sysctls, route metrics and setsockopts), this new mechanism provides some distinct advantages. Unlike sysctls, it can set parameters per connection. In contrast to route metrics, it can also use port numbers and information provided by a user level program. In addition, it could set parameters probabilistically for evaluation purposes (i.e. do something different on 10% of the flows and compare results with the other 90% of the flows). Also, in cases where IPv6 addresses contain geographic information, the rules to make changes based on the distance (or RTT) between the hosts are much easier than route metric rules and can be global. Finally, unlike setsockopt, it oes not require application changes and it can be updated easily at any time. Although the bpf cgroup framework already contains a sock related program type (BPF_PROG_TYPE_CGROUP_SOCK), I created the new type (BPF_PROG_TYPE_SOCK_OPS) beccause the existing type expects to be called only once during the connections's lifetime. In contrast, the new program type will be called multiple times from different places in the network stack code. For example, before sending SYN and SYN-ACKs to set an appropriate timeout, when the connection is established to set congestion control, etc. As a result it has "op" field to specify the type of operation requested. The purpose of this new program type is to simplify setting connection parameters, such as buffer sizes, TCP's SYN RTO, etc. For example, it is easy to use facebook's internal IPv6 addresses to determine if both hosts of a connection are in the same datacenter. Therefore, it is easy to write a BPF program to choose a small SYN RTO value when both hosts are in the same datacenter. This patch only contains the framework to support the new BPF program type, following patches add the functionality to set various connection parameters. This patch defines a new BPF program type: BPF_PROG_TYPE_SOCKET_OPS and a new bpf syscall command to load a new program of this type: BPF_PROG_LOAD_SOCKET_OPS. Two new corresponding structs (one for the kernel one for the user/BPF program): /* kernel version */ struct bpf_sock_ops_kern { struct sock *sk; __u32 op; union { __u32 reply; __u32 replylong[4]; }; }; /* user version * Some fields are in network byte order reflecting the sock struct * Use the bpf_ntohl helper macro in samples/bpf/bpf_endian.h to * convert them to host byte order. */ struct bpf_sock_ops { __u32 op; union { __u32 reply; __u32 replylong[4]; }; __u32 family; __u32 remote_ip4; /* In network byte order */ __u32 local_ip4; /* In network byte order */ __u32 remote_ip6[4]; /* In network byte order */ __u32 local_ip6[4]; /* In network byte order */ __u32 remote_port; /* In network byte order */ __u32 local_port; /* In host byte horder */ }; Currently there are two types of ops. The first type expects the BPF program to return a value which is then used by the caller (or a negative value to indicate the operation is not supported). The second type expects state changes to be done by the BPF program, for example through a setsockopt BPF helper function, and they ignore the return value. The reply fields of the bpf_sockt_ops struct are there in case a bpf program needs to return a value larger than an integer. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Daniel Borkmann <daniel@iogearbox.net> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-07-01 11:02:40 +08:00
extern struct inet_hashinfo tcp_hashinfo;
extern struct percpu_counter tcp_orphan_count;
void tcp_time_wait(struct sock *sk, int state, int timeo);
#define MAX_TCP_HEADER L1_CACHE_ALIGN(128 + MAX_HEADER)
#define MAX_TCP_OPTION_SPACE 40
#define TCP_MIN_SND_MSS 48
#define TCP_MIN_GSO_SIZE (TCP_MIN_SND_MSS - MAX_TCP_OPTION_SPACE)
/*
* Never offer a window over 32767 without using window scaling. Some
* poor stacks do signed 16bit maths!
*/
#define MAX_TCP_WINDOW 32767U
/* Minimal accepted MSS. It is (60+60+8) - (20+20). */
#define TCP_MIN_MSS 88U
/* The initial MTU to use for probing */
#define TCP_BASE_MSS 1024
/* probing interval, default to 10 minutes as per RFC4821 */
#define TCP_PROBE_INTERVAL 600
/* Specify interval when tcp mtu probing will stop */
#define TCP_PROBE_THRESHOLD 8
/* After receiving this amount of duplicate ACKs fast retransmit starts. */
#define TCP_FASTRETRANS_THRESH 3
/* Maximal number of ACKs sent quickly to accelerate slow-start. */
#define TCP_MAX_QUICKACKS 16U
/* Maximal number of window scale according to RFC1323 */
#define TCP_MAX_WSCALE 14U
/* urg_data states */
#define TCP_URG_VALID 0x0100
#define TCP_URG_NOTYET 0x0200
#define TCP_URG_READ 0x0400
#define TCP_RETR1 3 /*
* This is how many retries it does before it
* tries to figure out if the gateway is
* down. Minimal RFC value is 3; it corresponds
* to ~3sec-8min depending on RTO.
*/
#define TCP_RETR2 15 /*
* This should take at least
* 90 minutes to time out.
* RFC1122 says that the limit is 100 sec.
* 15 is ~13-30min depending on RTO.
*/
#define TCP_SYN_RETRIES 6 /* This is how many retries are done
* when active opening a connection.
* RFC1122 says the minimum retry MUST
* be at least 180secs. Nevertheless
* this value is corresponding to
* 63secs of retransmission with the
* current initial RTO.
*/
#define TCP_SYNACK_RETRIES 5 /* This is how may retries are done
* when passive opening a connection.
* This is corresponding to 31secs of
* retransmission with the current
* initial RTO.
*/
#define TCP_TIMEWAIT_LEN (60*HZ) /* how long to wait to destroy TIME-WAIT
* state, about 60 seconds */
#define TCP_FIN_TIMEOUT TCP_TIMEWAIT_LEN
/* BSD style FIN_WAIT2 deadlock breaker.
* It used to be 3min, new value is 60sec,
* to combine FIN-WAIT-2 timeout with
* TIME-WAIT timer.
*/
#define TCP_FIN_TIMEOUT_MAX (120 * HZ) /* max TCP_LINGER2 value (two minutes) */
#define TCP_DELACK_MAX ((unsigned)(HZ/5)) /* maximal time to delay before sending an ACK */
#if HZ >= 100
#define TCP_DELACK_MIN ((unsigned)(HZ/25)) /* minimal time to delay before sending an ACK */
#define TCP_ATO_MIN ((unsigned)(HZ/25))
#else
#define TCP_DELACK_MIN 4U
#define TCP_ATO_MIN 4U
#endif
#define TCP_RTO_MAX ((unsigned)(120*HZ))
#define TCP_RTO_MIN ((unsigned)(HZ/5))
#define TCP_TIMEOUT_MIN (2U) /* Min timeout for TCP timers in jiffies */
#define TCP_TIMEOUT_INIT ((unsigned)(1*HZ)) /* RFC6298 2.1 initial RTO value */
#define TCP_TIMEOUT_FALLBACK ((unsigned)(3*HZ)) /* RFC 1122 initial RTO value, now
* used as a fallback RTO for the
* initial data transmission if no
* valid RTT sample has been acquired,
* most likely due to retrans in 3WHS.
*/
#define TCP_RESOURCE_PROBE_INTERVAL ((unsigned)(HZ/2U)) /* Maximal interval between probes
* for local resources.
*/
#define TCP_KEEPALIVE_TIME (120*60*HZ) /* two hours */
#define TCP_KEEPALIVE_PROBES 9 /* Max of 9 keepalive probes */
#define TCP_KEEPALIVE_INTVL (75*HZ)
#define MAX_TCP_KEEPIDLE 32767
#define MAX_TCP_KEEPINTVL 32767
#define MAX_TCP_KEEPCNT 127
#define MAX_TCP_SYNCNT 127
#define TCP_SYNQ_INTERVAL (HZ/5) /* Period of SYNACK timer */
#define TCP_PAWS_24DAYS (60 * 60 * 24 * 24)
#define TCP_PAWS_MSL 60 /* Per-host timestamps are invalidated
* after this time. It should be equal
* (or greater than) TCP_TIMEWAIT_LEN
* to provide reliability equal to one
* provided by timewait state.
*/
#define TCP_PAWS_WINDOW 1 /* Replay window for per-host
* timestamps. It must be less than
* minimal timewait lifetime.
*/
/*
* TCP option
*/
#define TCPOPT_NOP 1 /* Padding */
#define TCPOPT_EOL 0 /* End of options */
#define TCPOPT_MSS 2 /* Segment size negotiating */
#define TCPOPT_WINDOW 3 /* Window scaling */
#define TCPOPT_SACK_PERM 4 /* SACK Permitted */
#define TCPOPT_SACK 5 /* SACK Block */
#define TCPOPT_TIMESTAMP 8 /* Better RTT estimations/PAWS */
#define TCPOPT_MD5SIG 19 /* MD5 Signature (RFC2385) */
#define TCPOPT_MPTCP 30 /* Multipath TCP (RFC6824) */
#define TCPOPT_FASTOPEN 34 /* Fast open (RFC7413) */
#define TCPOPT_EXP 254 /* Experimental */
/* Magic number to be after the option value for sharing TCP
* experimental options. See draft-ietf-tcpm-experimental-options-00.txt
*/
#define TCPOPT_FASTOPEN_MAGIC 0xF989
#define TCPOPT_SMC_MAGIC 0xE2D4C3D9
/*
* TCP option lengths
*/
#define TCPOLEN_MSS 4
#define TCPOLEN_WINDOW 3
#define TCPOLEN_SACK_PERM 2
#define TCPOLEN_TIMESTAMP 10
#define TCPOLEN_MD5SIG 18
#define TCPOLEN_FASTOPEN_BASE 2
#define TCPOLEN_EXP_FASTOPEN_BASE 4
#define TCPOLEN_EXP_SMC_BASE 6
/* But this is what stacks really send out. */
#define TCPOLEN_TSTAMP_ALIGNED 12
#define TCPOLEN_WSCALE_ALIGNED 4
#define TCPOLEN_SACKPERM_ALIGNED 4
#define TCPOLEN_SACK_BASE 2
#define TCPOLEN_SACK_BASE_ALIGNED 4
#define TCPOLEN_SACK_PERBLOCK 8
#define TCPOLEN_MD5SIG_ALIGNED 20
#define TCPOLEN_MSS_ALIGNED 4
#define TCPOLEN_EXP_SMC_BASE_ALIGNED 8
/* Flags in tp->nonagle */
#define TCP_NAGLE_OFF 1 /* Nagle's algo is disabled */
#define TCP_NAGLE_CORK 2 /* Socket is corked */
#define TCP_NAGLE_PUSH 4 /* Cork is overridden for already queued data */
/* TCP thin-stream limits */
#define TCP_THIN_LINEAR_RETRIES 6 /* After 6 linear retries, do exp. backoff */
/* TCP initial congestion window as per rfc6928 */
#define TCP_INIT_CWND 10
/* Bit Flags for sysctl_tcp_fastopen */
#define TFO_CLIENT_ENABLE 1
#define TFO_SERVER_ENABLE 2
#define TFO_CLIENT_NO_COOKIE 4 /* Data in SYN w/o cookie option */
/* Accept SYN data w/o any cookie option */
#define TFO_SERVER_COOKIE_NOT_REQD 0x200
/* Force enable TFO on all listeners, i.e., not requiring the
* TCP_FASTOPEN socket option.
*/
#define TFO_SERVER_WO_SOCKOPT1 0x400
/* sysctl variables for tcp */
extern int sysctl_tcp_max_orphans;
extern long sysctl_tcp_mem[3];
#define TCP_RACK_LOSS_DETECTION 0x1 /* Use RACK to detect losses */
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 07:38:48 +08:00
#define TCP_RACK_STATIC_REO_WND 0x2 /* Use static RACK reo wnd */
tcp: support DUPACK threshold in RACK This patch adds support for the classic DUPACK threshold rule (#DupThresh) in RACK. When the number of packets SACKed is greater or equal to the threshold, RACK sets the reordering window to zero which would immediately mark all the unsacked packets below the highest SACKed sequence lost. Since this approach is known to not work well with reordering, RACK only uses it if no reordering has been observed. The DUPACK threshold rule is a particularly useful extension to the fast recoveries triggered by RACK reordering timer. For example data-center transfers where the RTT is much smaller than a timer tick, or high RTT path where the default RTT/4 may take too long. Note that this patch differs slightly from RFC6675. RFC6675 considers a packet lost when at least #DupThresh higher-sequence packets are SACKed. With RACK, for connections that have seen reordering, RACK continues to use a dynamically-adaptive time-based reordering window to detect losses. But for connections on which we have not yet seen reordering, this patch considers a packet lost when at least one higher sequence packet is SACKed and the total number of SACKed packets is at least DupThresh. For example, suppose a connection has not seen reordering, and sends 10 packets, and packets 3, 5, 7 are SACKed. RFC6675 considers packets 1 and 2 lost. RACK considers packets 1, 2, 4, 6 lost. There is some small risk of spurious retransmits here due to reordering. However, this is mostly limited to the first flight of a connection on which the sender receives SACKs from reordering. And RFC 6675 and FACK loss detection have a similar risk on the first flight with reordering (it's just that the risk of spurious retransmits from reordering was slightly narrower for those older algorithms due to the margin of 3*MSS). Also the minimum reordering window is reduced from 1 msec to 0 to recover quicker on short RTT transfers. Therefore RACK is more aggressive in marking packets lost during recovery to reduce the reordering window timeouts. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Reviewed-by: Eric Dumazet <edumazet@google.com> Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com> Reviewed-by: Priyaranjan Jha <priyarjha@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2018-05-17 07:40:10 +08:00
#define TCP_RACK_NO_DUPTHRESH 0x4 /* Do not use DUPACK threshold in RACK */
extern atomic_long_t tcp_memory_allocated;
extern struct percpu_counter tcp_sockets_allocated;
extern unsigned long tcp_memory_pressure;
/* optimized version of sk_under_memory_pressure() for TCP sockets */
static inline bool tcp_under_memory_pressure(const struct sock *sk)
{
if (mem_cgroup_sockets_enabled && sk->sk_memcg &&
mem_cgroup_under_socket_pressure(sk->sk_memcg))
net: tcp_memcontrol: sanitize tcp memory accounting callbacks There won't be a tcp control soft limit, so integrating the memcg code into the global skmem limiting scheme complicates things unnecessarily. Replace this with simple and clear charge and uncharge calls--hidden behind a jump label--to account skb memory. Note that this is not purely aesthetic: as a result of shoehorning the per-memcg code into the same memory accounting functions that handle the global level, the old code would compare the per-memcg consumption against the smaller of the per-memcg limit and the global limit. This allowed the total consumption of multiple sockets to exceed the global limit, as long as the individual sockets stayed within bounds. After this change, the code will always compare the per-memcg consumption to the per-memcg limit, and the global consumption to the global limit, and thus close this loophole. Without a soft limit, the per-memcg memory pressure state in sockets is generally questionable. However, we did it until now, so we continue to enter it when the hard limit is hit, and packets are dropped, to let other sockets in the cgroup know that they shouldn't grow their transmit windows, either. However, keep it simple in the new callback model and leave memory pressure lazily when the next packet is accepted (as opposed to doing it synchroneously when packets are processed). When packets are dropped, network performance will already be in the toilet, so that should be a reasonable trade-off. As described above, consumption is now checked on the per-memcg level and the global level separately. Likewise, memory pressure states are maintained on both the per-memcg level and the global level, and a socket is considered under pressure when either level asserts as much. Signed-off-by: Johannes Weiner <hannes@cmpxchg.org> Reviewed-by: Vladimir Davydov <vdavydov@virtuozzo.com> Acked-by: David S. Miller <davem@davemloft.net> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2016-01-15 07:21:14 +08:00
return true;
return READ_ONCE(tcp_memory_pressure);
}
/*
* The next routines deal with comparing 32 bit unsigned ints
* and worry about wraparound (automatic with unsigned arithmetic).
*/
static inline bool before(__u32 seq1, __u32 seq2)
{
return (__s32)(seq1-seq2) < 0;
}
#define after(seq2, seq1) before(seq1, seq2)
/* is s2<=s1<=s3 ? */
static inline bool between(__u32 seq1, __u32 seq2, __u32 seq3)
{
return seq3 - seq2 >= seq1 - seq2;
}
static inline bool tcp_out_of_memory(struct sock *sk)
{
if (sk->sk_wmem_queued > SOCK_MIN_SNDBUF &&
sk_memory_allocated(sk) > sk_prot_mem_limits(sk, 2))
return true;
return false;
}
void sk_forced_mem_schedule(struct sock *sk, int size);
static inline bool tcp_too_many_orphans(struct sock *sk, int shift)
{
struct percpu_counter *ocp = sk->sk_prot->orphan_count;
int orphans = percpu_counter_read_positive(ocp);
if (orphans << shift > sysctl_tcp_max_orphans) {
orphans = percpu_counter_sum_positive(ocp);
if (orphans << shift > sysctl_tcp_max_orphans)
return true;
}
return false;
}
bool tcp_check_oom(struct sock *sk, int shift);
extern struct proto tcp_prot;
#define TCP_INC_STATS(net, field) SNMP_INC_STATS((net)->mib.tcp_statistics, field)
#define __TCP_INC_STATS(net, field) __SNMP_INC_STATS((net)->mib.tcp_statistics, field)
#define TCP_DEC_STATS(net, field) SNMP_DEC_STATS((net)->mib.tcp_statistics, field)
#define TCP_ADD_STATS(net, field, val) SNMP_ADD_STATS((net)->mib.tcp_statistics, field, val)
void tcp_tasklet_init(void);
int tcp_v4_err(struct sk_buff *skb, u32);
void tcp_shutdown(struct sock *sk, int how);
int tcp_v4_early_demux(struct sk_buff *skb);
int tcp_v4_rcv(struct sk_buff *skb);
int tcp_v4_tw_remember_stamp(struct inet_timewait_sock *tw);
int tcp_sendmsg(struct sock *sk, struct msghdr *msg, size_t size);
int tcp_sendmsg_locked(struct sock *sk, struct msghdr *msg, size_t size);
int tcp_sendpage(struct sock *sk, struct page *page, int offset, size_t size,
int flags);
int tcp_sendpage_locked(struct sock *sk, struct page *page, int offset,
size_t size, int flags);
ssize_t do_tcp_sendpages(struct sock *sk, struct page *page, int offset,
size_t size, int flags);
int tcp_send_mss(struct sock *sk, int *size_goal, int flags);
void tcp_push(struct sock *sk, int flags, int mss_now, int nonagle,
int size_goal);
void tcp_release_cb(struct sock *sk);
void tcp_wfree(struct sk_buff *skb);
void tcp_write_timer_handler(struct sock *sk);
void tcp_delack_timer_handler(struct sock *sk);
int tcp_ioctl(struct sock *sk, int cmd, unsigned long arg);
int tcp_rcv_state_process(struct sock *sk, struct sk_buff *skb);
void tcp_rcv_established(struct sock *sk, struct sk_buff *skb);
void tcp_rcv_space_adjust(struct sock *sk);
int tcp_twsk_unique(struct sock *sk, struct sock *sktw, void *twp);
void tcp_twsk_destructor(struct sock *sk);
ssize_t tcp_splice_read(struct socket *sk, loff_t *ppos,
struct pipe_inode_info *pipe, size_t len,
unsigned int flags);
tcp: do not delay ACK in DCTCP upon CE status change Per DCTCP RFC8257 (Section 3.2) the ACK reflecting the CE status change has to be sent immediately so the sender can respond quickly: """ When receiving packets, the CE codepoint MUST be processed as follows: 1. If the CE codepoint is set and DCTCP.CE is false, set DCTCP.CE to true and send an immediate ACK. 2. If the CE codepoint is not set and DCTCP.CE is true, set DCTCP.CE to false and send an immediate ACK. """ Previously DCTCP implementation may continue to delay the ACK. This patch fixes that to implement the RFC by forcing an immediate ACK. Tested with this packetdrill script provided by Larry Brakmo 0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 0.000 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 0.000 setsockopt(3, SOL_TCP, TCP_CONGESTION, "dctcp", 5) = 0 0.000 bind(3, ..., ...) = 0 0.000 listen(3, 1) = 0 0.100 < [ect0] SEW 0:0(0) win 32792 <mss 1000,sackOK,nop,nop,nop,wscale 7> 0.100 > SE. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 8> 0.110 < [ect0] . 1:1(0) ack 1 win 257 0.200 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_SOCKET, SO_DEBUG, [1], 4) = 0 0.200 < [ect0] . 1:1001(1000) ack 1 win 257 0.200 > [ect01] . 1:1(0) ack 1001 0.200 write(4, ..., 1) = 1 0.200 > [ect01] P. 1:2(1) ack 1001 0.200 < [ect0] . 1001:2001(1000) ack 2 win 257 +0.005 < [ce] . 2001:3001(1000) ack 2 win 257 +0.000 > [ect01] . 2:2(0) ack 2001 // Previously the ACK below would be delayed by 40ms +0.000 > [ect01] E. 2:2(0) ack 3001 +0.500 < F. 9501:9501(0) ack 4 win 257 Signed-off-by: Yuchung Cheng <ycheng@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2018-07-19 04:56:36 +08:00
void tcp_enter_quickack_mode(struct sock *sk, unsigned int max_quickacks);
static inline void tcp_dec_quickack_mode(struct sock *sk,
const unsigned int pkts)
{
struct inet_connection_sock *icsk = inet_csk(sk);
if (icsk->icsk_ack.quick) {
if (pkts >= icsk->icsk_ack.quick) {
icsk->icsk_ack.quick = 0;
/* Leaving quickack mode we deflate ATO. */
icsk->icsk_ack.ato = TCP_ATO_MIN;
} else
icsk->icsk_ack.quick -= pkts;
}
}
#define TCP_ECN_OK 1
#define TCP_ECN_QUEUE_CWR 2
#define TCP_ECN_DEMAND_CWR 4
#define TCP_ECN_SEEN 8
enum tcp_tw_status {
TCP_TW_SUCCESS = 0,
TCP_TW_RST = 1,
TCP_TW_ACK = 2,
TCP_TW_SYN = 3
};
enum tcp_tw_status tcp_timewait_state_process(struct inet_timewait_sock *tw,
struct sk_buff *skb,
const struct tcphdr *th);
struct sock *tcp_check_req(struct sock *sk, struct sk_buff *skb,
struct request_sock *req, bool fastopen,
bool *lost_race);
int tcp_child_process(struct sock *parent, struct sock *child,
struct sk_buff *skb);
tcp: reduce spurious retransmits due to transient SACK reneging This commit reduces spurious retransmits due to apparent SACK reneging by only reacting to SACK reneging that persists for a short delay. When a sequence space hole at snd_una is filled, some TCP receivers send a series of ACKs as they apparently scan their out-of-order queue and cumulatively ACK all the packets that have now been consecutiveyly received. This is essentially misbehavior B in "Misbehaviors in TCP SACK generation" ACM SIGCOMM Computer Communication Review, April 2011, so we suspect that this is from several common OSes (Windows 2000, Windows Server 2003, Windows XP). However, this issue has also been seen in other cases, e.g. the netdev thread "TCP being hoodwinked into spurious retransmissions by lack of timestamps?" from March 2014, where the receiver was thought to be a BSD box. Since snd_una would temporarily be adjacent to a previously SACKed range in these scenarios, this receiver behavior triggered the Linux SACK reneging code path in the sender. This led the sender to clear the SACK scoreboard, enter CA_Loss, and spuriously retransmit (potentially) every packet from the entire write queue at line rate just a few milliseconds before the ACK for each packet arrives at the sender. To avoid such situations, now when a sender sees apparent reneging it does not yet retransmit, but rather adjusts the RTO timer to give the receiver a little time (max(RTT/2, 10ms)) to send us some more ACKs that will restore sanity to the SACK scoreboard. If the reneging persists until this RTO then, as before, we clear the SACK scoreboard and enter CA_Loss. A 10ms delay tolerates a receiver sending such a stream of ACKs at 56Kbit/sec. And to allow for receivers with slower or more congested paths, we wait for at least RTT/2. We validated the resulting max(RTT/2, 10ms) delay formula with a mix of North American and South American Google web server traffic, and found that for ACKs displaying transient reneging: (1) 90% of inter-ACK delays were less than 10ms (2) 99% of inter-ACK delays were less than RTT/2 In tests on Google web servers this commit reduced reneging events by 75%-90% (as measured by the TcpExtTCPSACKReneging counter), without any measurable impact on latency for user HTTP and SPDY requests. Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-08-05 07:12:29 +08:00
void tcp_enter_loss(struct sock *sk);
void tcp_cwnd_reduction(struct sock *sk, int newly_acked_sacked, int newly_lost, int flag);
void tcp_clear_retrans(struct tcp_sock *tp);
void tcp_update_metrics(struct sock *sk);
void tcp_init_metrics(struct sock *sk);
void tcp_metrics_init(void);
bool tcp_peer_is_proven(struct request_sock *req, struct dst_entry *dst);
void tcp_close(struct sock *sk, long timeout);
void tcp_init_sock(struct sock *sk);
void tcp_init_transfer(struct sock *sk, int bpf_op, struct sk_buff *skb);
__poll_t tcp_poll(struct file *file, struct socket *sock,
struct poll_table_struct *wait);
int tcp_getsockopt(struct sock *sk, int level, int optname,
char __user *optval, int __user *optlen);
int tcp_setsockopt(struct sock *sk, int level, int optname, sockptr_t optval,
unsigned int optlen);
void tcp_set_keepalive(struct sock *sk, int val);
void tcp_syn_ack_timeout(const struct request_sock *req);
int tcp_recvmsg(struct sock *sk, struct msghdr *msg, size_t len, int nonblock,
int flags, int *addr_len);
int tcp_set_rcvlowat(struct sock *sk, int val);
void tcp_data_ready(struct sock *sk);
#ifdef CONFIG_MMU
tcp: implement mmap() for zero copy receive Some networks can make sure TCP payload can exactly fit 4KB pages, with well chosen MSS/MTU and architectures. Implement mmap() system call so that applications can avoid copying data without complex splice() games. Note that a successful mmap( X bytes) on TCP socket is consuming bytes, as if recvmsg() has been done. (tp->copied += X) Only PROT_READ mappings are accepted, as skb page frags are fundamentally shared and read only. If tcp_mmap() finds data that is not a full page, or a patch of urgent data, -EINVAL is returned, no bytes are consumed. Application must fallback to recvmsg() to read the problematic sequence. mmap() wont block, regardless of socket being in blocking or non-blocking mode. If not enough bytes are in receive queue, mmap() would return -EAGAIN, or -EIO if socket is in a state where no other bytes can be added into receive queue. An application might use SO_RCVLOWAT, poll() and/or ioctl( FIONREAD) to efficiently use mmap() On the sender side, MSG_EOR might help to clearly separate unaligned headers and 4K-aligned chunks if necessary. Tested: mlx4 (cx-3) 40Gbit NIC, with tcp_mmap program provided in following patch. MTU set to 4168 (4096 TCP payload, 40 bytes IPv6 header, 32 bytes TCP header) Without mmap() (tcp_mmap -s) received 32768 MB (0 % mmap'ed) in 8.13342 s, 33.7961 Gbit, cpu usage user:0.034 sys:3.778, 116.333 usec per MB, 63062 c-switches received 32768 MB (0 % mmap'ed) in 8.14501 s, 33.748 Gbit, cpu usage user:0.029 sys:3.997, 122.864 usec per MB, 61903 c-switches received 32768 MB (0 % mmap'ed) in 8.11723 s, 33.8635 Gbit, cpu usage user:0.048 sys:3.964, 122.437 usec per MB, 62983 c-switches received 32768 MB (0 % mmap'ed) in 8.39189 s, 32.7552 Gbit, cpu usage user:0.038 sys:4.181, 128.754 usec per MB, 55834 c-switches With mmap() on receiver (tcp_mmap -s -z) received 32768 MB (100 % mmap'ed) in 8.03083 s, 34.2278 Gbit, cpu usage user:0.024 sys:1.466, 45.4712 usec per MB, 65479 c-switches received 32768 MB (100 % mmap'ed) in 7.98805 s, 34.4111 Gbit, cpu usage user:0.026 sys:1.401, 43.5486 usec per MB, 65447 c-switches received 32768 MB (100 % mmap'ed) in 7.98377 s, 34.4296 Gbit, cpu usage user:0.028 sys:1.452, 45.166 usec per MB, 65496 c-switches received 32768 MB (99.9969 % mmap'ed) in 8.01838 s, 34.281 Gbit, cpu usage user:0.02 sys:1.446, 44.7388 usec per MB, 65505 c-switches Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2018-04-17 01:33:38 +08:00
int tcp_mmap(struct file *file, struct socket *sock,
struct vm_area_struct *vma);
#endif
void tcp_parse_options(const struct net *net, const struct sk_buff *skb,
struct tcp_options_received *opt_rx,
int estab, struct tcp_fastopen_cookie *foc);
const u8 *tcp_parse_md5sig_option(const struct tcphdr *th);
/*
* BPF SKB-less helpers
*/
u16 tcp_v4_get_syncookie(struct sock *sk, struct iphdr *iph,
struct tcphdr *th, u32 *cookie);
u16 tcp_v6_get_syncookie(struct sock *sk, struct ipv6hdr *iph,
struct tcphdr *th, u32 *cookie);
u16 tcp_get_syncookie_mss(struct request_sock_ops *rsk_ops,
const struct tcp_request_sock_ops *af_ops,
struct sock *sk, struct tcphdr *th);
/*
* TCP v4 functions exported for the inet6 API
*/
void tcp_v4_send_check(struct sock *sk, struct sk_buff *skb);
void tcp_v4_mtu_reduced(struct sock *sk);
void tcp_req_err(struct sock *sk, u32 seq, bool abort);
void tcp_ld_RTO_revert(struct sock *sk, u32 seq);
int tcp_v4_conn_request(struct sock *sk, struct sk_buff *skb);
struct sock *tcp_create_openreq_child(const struct sock *sk,
struct request_sock *req,
struct sk_buff *skb);
net: tcp: add per route congestion control This work adds the possibility to define a per route/destination congestion control algorithm. Generally, this opens up the possibility for a machine with different links to enforce specific congestion control algorithms with optimal strategies for each of them based on their network characteristics, even transparently for a single application listening on all links. For our specific use case, this additionally facilitates deployment of DCTCP, for example, applications can easily serve internal traffic/dsts in DCTCP and external one with CUBIC. Other scenarios would also allow for utilizing e.g. long living, low priority background flows for certain destinations/routes while still being able for normal traffic to utilize the default congestion control algorithm. We also thought about a per netns setting (where different defaults are possible), but given its actually a link specific property, we argue that a per route/destination setting is the most natural and flexible. The administrator can utilize this through ip-route(8) by appending "congctl [lock] <name>", where <name> denotes the name of a congestion control algorithm and the optional lock parameter allows to enforce the given algorithm so that applications in user space would not be allowed to overwrite that algorithm for that destination. The dst metric lookups are being done when a dst entry is already available in order to avoid a costly lookup and still before the algorithms are being initialized, thus overhead is very low when the feature is not being used. While the client side would need to drop the current reference on the module, on server side this can actually even be avoided as we just got a flat-copied socket clone. Joint work with Florian Westphal. Suggested-by: Hannes Frederic Sowa <hannes@stressinduktion.org> Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: Daniel Borkmann <dborkman@redhat.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:48 +08:00
void tcp_ca_openreq_child(struct sock *sk, const struct dst_entry *dst);
struct sock *tcp_v4_syn_recv_sock(const struct sock *sk, struct sk_buff *skb,
struct request_sock *req,
struct dst_entry *dst,
struct request_sock *req_unhash,
bool *own_req);
int tcp_v4_do_rcv(struct sock *sk, struct sk_buff *skb);
int tcp_v4_connect(struct sock *sk, struct sockaddr *uaddr, int addr_len);
int tcp_connect(struct sock *sk);
enum tcp_synack_type {
TCP_SYNACK_NORMAL,
TCP_SYNACK_FASTOPEN,
TCP_SYNACK_COOKIE,
};
struct sk_buff *tcp_make_synack(const struct sock *sk, struct dst_entry *dst,
struct request_sock *req,
struct tcp_fastopen_cookie *foc,
bpf: tcp: Add bpf_skops_hdr_opt_len() and bpf_skops_write_hdr_opt() The bpf prog needs to parse the SYN header to learn what options have been sent by the peer's bpf-prog before writing its options into SYNACK. This patch adds a "syn_skb" arg to tcp_make_synack() and send_synack(). This syn_skb will eventually be made available (as read-only) to the bpf prog. This will be the only SYN packet available to the bpf prog during syncookie. For other regular cases, the bpf prog can also use the saved_syn. When writing options, the bpf prog will first be called to tell the kernel its required number of bytes. It is done by the new bpf_skops_hdr_opt_len(). The bpf prog will only be called when the new BPF_SOCK_OPS_WRITE_HDR_OPT_CB_FLAG is set in tp->bpf_sock_ops_cb_flags. When the bpf prog returns, the kernel will know how many bytes are needed and then update the "*remaining" arg accordingly. 4 byte alignment will be included in the "*remaining" before this function returns. The 4 byte aligned number of bytes will also be stored into the opts->bpf_opt_len. "bpf_opt_len" is a newly added member to the struct tcp_out_options. Then the new bpf_skops_write_hdr_opt() will call the bpf prog to write the header options. The bpf prog is only called if it has reserved spaces before (opts->bpf_opt_len > 0). The bpf prog is the last one getting a chance to reserve header space and writing the header option. These two functions are half implemented to highlight the changes in TCP stack. The actual codes preparing the bpf running context and invoking the bpf prog will be added in the later patch with other necessary bpf pieces. Signed-off-by: Martin KaFai Lau <kafai@fb.com> Signed-off-by: Alexei Starovoitov <ast@kernel.org> Reviewed-by: Eric Dumazet <edumazet@google.com> Link: https://lore.kernel.org/bpf/20200820190052.2885316-1-kafai@fb.com
2020-08-21 03:00:52 +08:00
enum tcp_synack_type synack_type,
struct sk_buff *syn_skb);
int tcp_disconnect(struct sock *sk, int flags);
void tcp_finish_connect(struct sock *sk, struct sk_buff *skb);
int tcp_send_rcvq(struct sock *sk, struct msghdr *msg, size_t size);
net: tcp: ipv6_mapped needs sk_rx_dst_set method commit 5d299f3d3c8a2fb (net: ipv6: fix TCP early demux) added a regression for ipv6_mapped case. [ 67.422369] SELinux: initialized (dev autofs, type autofs), uses genfs_contexts [ 67.449678] SELinux: initialized (dev autofs, type autofs), uses genfs_contexts [ 92.631060] BUG: unable to handle kernel NULL pointer dereference at (null) [ 92.631435] IP: [< (null)>] (null) [ 92.631645] PGD 0 [ 92.631846] Oops: 0010 [#1] SMP [ 92.632095] Modules linked in: autofs4 sunrpc ipv6 dm_mirror dm_region_hash dm_log dm_multipath dm_mod video sbs sbshc battery ac lp parport sg snd_hda_intel snd_hda_codec snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device pcspkr snd_pcm_oss snd_mixer_oss snd_pcm snd_timer serio_raw button floppy snd i2c_i801 i2c_core soundcore snd_page_alloc shpchp ide_cd_mod cdrom microcode ehci_hcd ohci_hcd uhci_hcd [ 92.634294] CPU 0 [ 92.634294] Pid: 4469, comm: sendmail Not tainted 3.6.0-rc1 #3 [ 92.634294] RIP: 0010:[<0000000000000000>] [< (null)>] (null) [ 92.634294] RSP: 0018:ffff880245fc7cb0 EFLAGS: 00010282 [ 92.634294] RAX: ffffffffa01985f0 RBX: ffff88024827ad00 RCX: 0000000000000000 [ 92.634294] RDX: 0000000000000218 RSI: ffff880254735380 RDI: ffff88024827ad00 [ 92.634294] RBP: ffff880245fc7cc8 R08: 0000000000000001 R09: 0000000000000000 [ 92.634294] R10: 0000000000000000 R11: ffff880245fc7bf8 R12: ffff880254735380 [ 92.634294] R13: ffff880254735380 R14: 0000000000000000 R15: 7fffffffffff0218 [ 92.634294] FS: 00007f4516ccd6f0(0000) GS:ffff880256600000(0000) knlGS:0000000000000000 [ 92.634294] CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b [ 92.634294] CR2: 0000000000000000 CR3: 0000000245ed1000 CR4: 00000000000007f0 [ 92.634294] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 92.634294] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 92.634294] Process sendmail (pid: 4469, threadinfo ffff880245fc6000, task ffff880254b8cac0) [ 92.634294] Stack: [ 92.634294] ffffffff813837a7 ffff88024827ad00 ffff880254b6b0e8 ffff880245fc7d68 [ 92.634294] ffffffff81385083 00000000001d2680 ffff8802547353a8 ffff880245fc7d18 [ 92.634294] ffffffff8105903a ffff88024827ad60 0000000000000002 00000000000000ff [ 92.634294] Call Trace: [ 92.634294] [<ffffffff813837a7>] ? tcp_finish_connect+0x2c/0xfa [ 92.634294] [<ffffffff81385083>] tcp_rcv_state_process+0x2b6/0x9c6 [ 92.634294] [<ffffffff8105903a>] ? sched_clock_cpu+0xc3/0xd1 [ 92.634294] [<ffffffff81059073>] ? local_clock+0x2b/0x3c [ 92.634294] [<ffffffff8138caf3>] tcp_v4_do_rcv+0x63a/0x670 [ 92.634294] [<ffffffff8133278e>] release_sock+0x128/0x1bd [ 92.634294] [<ffffffff8139f060>] __inet_stream_connect+0x1b1/0x352 [ 92.634294] [<ffffffff813325f5>] ? lock_sock_nested+0x74/0x7f [ 92.634294] [<ffffffff8104b333>] ? wake_up_bit+0x25/0x25 [ 92.634294] [<ffffffff813325f5>] ? lock_sock_nested+0x74/0x7f [ 92.634294] [<ffffffff8139f223>] ? inet_stream_connect+0x22/0x4b [ 92.634294] [<ffffffff8139f234>] inet_stream_connect+0x33/0x4b [ 92.634294] [<ffffffff8132e8cf>] sys_connect+0x78/0x9e [ 92.634294] [<ffffffff813fd407>] ? sysret_check+0x1b/0x56 [ 92.634294] [<ffffffff81088503>] ? __audit_syscall_entry+0x195/0x1c8 [ 92.634294] [<ffffffff811cc26e>] ? trace_hardirqs_on_thunk+0x3a/0x3f [ 92.634294] [<ffffffff813fd3e2>] system_call_fastpath+0x16/0x1b [ 92.634294] Code: Bad RIP value. [ 92.634294] RIP [< (null)>] (null) [ 92.634294] RSP <ffff880245fc7cb0> [ 92.634294] CR2: 0000000000000000 [ 92.648982] ---[ end trace 24e2bed94314c8d9 ]--- [ 92.649146] Kernel panic - not syncing: Fatal exception in interrupt Fix this using inet_sk_rx_dst_set(), and export this function in case IPv6 is modular. Reported-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2012-08-09 22:11:00 +08:00
void inet_sk_rx_dst_set(struct sock *sk, const struct sk_buff *skb);
/* From syncookies.c */
struct sock *tcp_get_cookie_sock(struct sock *sk, struct sk_buff *skb,
struct request_sock *req,
struct dst_entry *dst, u32 tsoff);
int __cookie_v4_check(const struct iphdr *iph, const struct tcphdr *th,
u32 cookie);
struct sock *cookie_v4_check(struct sock *sk, struct sk_buff *skb);
struct request_sock *cookie_tcp_reqsk_alloc(const struct request_sock_ops *ops,
struct sock *sk, struct sk_buff *skb);
#ifdef CONFIG_SYN_COOKIES
/* Syncookies use a monotonic timer which increments every 60 seconds.
* This counter is used both as a hash input and partially encoded into
* the cookie value. A cookie is only validated further if the delta
* between the current counter value and the encoded one is less than this,
* i.e. a sent cookie is valid only at most for 2*60 seconds (or less if
* the counter advances immediately after a cookie is generated).
*/
#define MAX_SYNCOOKIE_AGE 2
#define TCP_SYNCOOKIE_PERIOD (60 * HZ)
#define TCP_SYNCOOKIE_VALID (MAX_SYNCOOKIE_AGE * TCP_SYNCOOKIE_PERIOD)
/* syncookies: remember time of last synqueue overflow
* But do not dirty this field too often (once per second is enough)
* It is racy as we do not hold a lock, but race is very minor.
*/
static inline void tcp_synq_overflow(const struct sock *sk)
{
tcp: Avoid TCP syncookie rejected by SO_REUSEPORT socket Although the actual cookie check "__cookie_v[46]_check()" does not involve sk specific info, it checks whether the sk has recent synq overflow event in "tcp_synq_no_recent_overflow()". The tcp_sk(sk)->rx_opt.ts_recent_stamp is updated every second when it has sent out a syncookie (through "tcp_synq_overflow()"). The above per sk "recent synq overflow event timestamp" works well for non SO_REUSEPORT use case. However, it may cause random connection request reject/discard when SO_REUSEPORT is used with syncookie because it fails the "tcp_synq_no_recent_overflow()" test. When SO_REUSEPORT is used, it usually has multiple listening socks serving TCP connection requests destinated to the same local IP:PORT. There are cases that the TCP-ACK-COOKIE may not be received by the same sk that sent out the syncookie. For example, if reuse->socks[] began with {sk0, sk1}, 1) sk1 sent out syncookies and tcp_sk(sk1)->rx_opt.ts_recent_stamp was updated. 2) the reuse->socks[] became {sk1, sk2} later. e.g. sk0 was first closed and then sk2 was added. Here, sk2 does not have ts_recent_stamp set. There are other ordering that will trigger the similar situation below but the idea is the same. 3) When the TCP-ACK-COOKIE comes back, sk2 was selected. "tcp_synq_no_recent_overflow(sk2)" returns true. In this case, all syncookies sent by sk1 will be handled (and rejected) by sk2 while sk1 is still alive. The userspace may create and remove listening SO_REUSEPORT sockets as it sees fit. E.g. Adding new thread (and SO_REUSEPORT sock) to handle incoming requests, old process stopping and new process starting...etc. With or without SO_ATTACH_REUSEPORT_[CB]BPF, the sockets leaving and joining a reuseport group makes picking the same sk to check the syncookie very difficult (if not impossible). The later patches will allow bpf prog more flexibility in deciding where a sk should be located in a bpf map and selecting a particular SO_REUSEPORT sock as it sees fit. e.g. Without closing any sock, replace the whole bpf reuseport_array in one map_update() by using map-in-map. Getting the syncookie check working smoothly across socks in the same "reuse->socks[]" is important. A partial solution is to set the newly added sk's ts_recent_stamp to the max ts_recent_stamp of a reuseport group but that will require to iterate through reuse->socks[] OR pessimistically set it to "now - TCP_SYNCOOKIE_VALID" when a sk is joining a reuseport group. However, neither of them will solve the existing sk getting moved around the reuse->socks[] and that sk may not have ts_recent_stamp updated, unlikely under continuous synflood but not impossible. This patch opts to treat the reuseport group as a whole when considering the last synq overflow timestamp since they are serving the same IP:PORT from the userspace (and BPF program) perspective. "synq_overflow_ts" is added to "struct sock_reuseport". The tcp_synq_overflow() and tcp_synq_no_recent_overflow() will update/check reuse->synq_overflow_ts if the sk is in a reuseport group. Similar to the reuseport decision in __inet_lookup_listener(), both sk->sk_reuseport and sk->sk_reuseport_cb are tested for SO_REUSEPORT usage. Update on "synq_overflow_ts" happens at roughly once every second. A synflood test was done with a 16 rx-queues and 16 reuseport sockets. No meaningful performance change is observed. Before and after the change is ~9Mpps in IPv4. Cc: Eric Dumazet <edumazet@google.com> Signed-off-by: Martin KaFai Lau <kafai@fb.com> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
2018-08-08 16:01:21 +08:00
unsigned int last_overflow;
unsigned int now = jiffies;
tcp: Avoid TCP syncookie rejected by SO_REUSEPORT socket Although the actual cookie check "__cookie_v[46]_check()" does not involve sk specific info, it checks whether the sk has recent synq overflow event in "tcp_synq_no_recent_overflow()". The tcp_sk(sk)->rx_opt.ts_recent_stamp is updated every second when it has sent out a syncookie (through "tcp_synq_overflow()"). The above per sk "recent synq overflow event timestamp" works well for non SO_REUSEPORT use case. However, it may cause random connection request reject/discard when SO_REUSEPORT is used with syncookie because it fails the "tcp_synq_no_recent_overflow()" test. When SO_REUSEPORT is used, it usually has multiple listening socks serving TCP connection requests destinated to the same local IP:PORT. There are cases that the TCP-ACK-COOKIE may not be received by the same sk that sent out the syncookie. For example, if reuse->socks[] began with {sk0, sk1}, 1) sk1 sent out syncookies and tcp_sk(sk1)->rx_opt.ts_recent_stamp was updated. 2) the reuse->socks[] became {sk1, sk2} later. e.g. sk0 was first closed and then sk2 was added. Here, sk2 does not have ts_recent_stamp set. There are other ordering that will trigger the similar situation below but the idea is the same. 3) When the TCP-ACK-COOKIE comes back, sk2 was selected. "tcp_synq_no_recent_overflow(sk2)" returns true. In this case, all syncookies sent by sk1 will be handled (and rejected) by sk2 while sk1 is still alive. The userspace may create and remove listening SO_REUSEPORT sockets as it sees fit. E.g. Adding new thread (and SO_REUSEPORT sock) to handle incoming requests, old process stopping and new process starting...etc. With or without SO_ATTACH_REUSEPORT_[CB]BPF, the sockets leaving and joining a reuseport group makes picking the same sk to check the syncookie very difficult (if not impossible). The later patches will allow bpf prog more flexibility in deciding where a sk should be located in a bpf map and selecting a particular SO_REUSEPORT sock as it sees fit. e.g. Without closing any sock, replace the whole bpf reuseport_array in one map_update() by using map-in-map. Getting the syncookie check working smoothly across socks in the same "reuse->socks[]" is important. A partial solution is to set the newly added sk's ts_recent_stamp to the max ts_recent_stamp of a reuseport group but that will require to iterate through reuse->socks[] OR pessimistically set it to "now - TCP_SYNCOOKIE_VALID" when a sk is joining a reuseport group. However, neither of them will solve the existing sk getting moved around the reuse->socks[] and that sk may not have ts_recent_stamp updated, unlikely under continuous synflood but not impossible. This patch opts to treat the reuseport group as a whole when considering the last synq overflow timestamp since they are serving the same IP:PORT from the userspace (and BPF program) perspective. "synq_overflow_ts" is added to "struct sock_reuseport". The tcp_synq_overflow() and tcp_synq_no_recent_overflow() will update/check reuse->synq_overflow_ts if the sk is in a reuseport group. Similar to the reuseport decision in __inet_lookup_listener(), both sk->sk_reuseport and sk->sk_reuseport_cb are tested for SO_REUSEPORT usage. Update on "synq_overflow_ts" happens at roughly once every second. A synflood test was done with a 16 rx-queues and 16 reuseport sockets. No meaningful performance change is observed. Before and after the change is ~9Mpps in IPv4. Cc: Eric Dumazet <edumazet@google.com> Signed-off-by: Martin KaFai Lau <kafai@fb.com> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
2018-08-08 16:01:21 +08:00
if (sk->sk_reuseport) {
struct sock_reuseport *reuse;
reuse = rcu_dereference(sk->sk_reuseport_cb);
if (likely(reuse)) {
last_overflow = READ_ONCE(reuse->synq_overflow_ts);
tcp: fix rejected syncookies due to stale timestamps If no synflood happens for a long enough period of time, then the synflood timestamp isn't refreshed and jiffies can advance so much that time_after32() can't accurately compare them any more. Therefore, we can end up in a situation where time_after32(now, last_overflow + HZ) returns false, just because these two values are too far apart. In that case, the synflood timestamp isn't updated as it should be, which can trick tcp_synq_no_recent_overflow() into rejecting valid syncookies. For example, let's consider the following scenario on a system with HZ=1000: * The synflood timestamp is 0, either because that's the timestamp of the last synflood or, more commonly, because we're working with a freshly created socket. * We receive a new SYN, which triggers synflood protection. Let's say that this happens when jiffies == 2147484649 (that is, 'synflood timestamp' + HZ + 2^31 + 1). * Then tcp_synq_overflow() doesn't update the synflood timestamp, because time_after32(2147484649, 1000) returns false. With: - 2147484649: the value of jiffies, aka. 'now'. - 1000: the value of 'last_overflow' + HZ. * A bit later, we receive the ACK completing the 3WHS. But cookie_v[46]_check() rejects it because tcp_synq_no_recent_overflow() says that we're not under synflood. That's because time_after32(2147484649, 120000) returns false. With: - 2147484649: the value of jiffies, aka. 'now'. - 120000: the value of 'last_overflow' + TCP_SYNCOOKIE_VALID. Of course, in reality jiffies would have increased a bit, but this condition will last for the next 119 seconds, which is far enough to accommodate for jiffie's growth. Fix this by updating the overflow timestamp whenever jiffies isn't within the [last_overflow, last_overflow + HZ] range. That shouldn't have any performance impact since the update still happens at most once per second. Now we're guaranteed to have fresh timestamps while under synflood, so tcp_synq_no_recent_overflow() can safely use it with time_after32() in such situations. Stale timestamps can still make tcp_synq_no_recent_overflow() return the wrong verdict when not under synflood. This will be handled in the next patch. For 64 bits architectures, the problem was introduced with the conversion of ->tw_ts_recent_stamp to 32 bits integer by commit cca9bab1b72c ("tcp: use monotonic timestamps for PAWS"). The problem has always been there on 32 bits architectures. Fixes: cca9bab1b72c ("tcp: use monotonic timestamps for PAWS") Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Guillaume Nault <gnault@redhat.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2019-12-06 19:38:36 +08:00
if (!time_between32(now, last_overflow,
last_overflow + HZ))
tcp: Avoid TCP syncookie rejected by SO_REUSEPORT socket Although the actual cookie check "__cookie_v[46]_check()" does not involve sk specific info, it checks whether the sk has recent synq overflow event in "tcp_synq_no_recent_overflow()". The tcp_sk(sk)->rx_opt.ts_recent_stamp is updated every second when it has sent out a syncookie (through "tcp_synq_overflow()"). The above per sk "recent synq overflow event timestamp" works well for non SO_REUSEPORT use case. However, it may cause random connection request reject/discard when SO_REUSEPORT is used with syncookie because it fails the "tcp_synq_no_recent_overflow()" test. When SO_REUSEPORT is used, it usually has multiple listening socks serving TCP connection requests destinated to the same local IP:PORT. There are cases that the TCP-ACK-COOKIE may not be received by the same sk that sent out the syncookie. For example, if reuse->socks[] began with {sk0, sk1}, 1) sk1 sent out syncookies and tcp_sk(sk1)->rx_opt.ts_recent_stamp was updated. 2) the reuse->socks[] became {sk1, sk2} later. e.g. sk0 was first closed and then sk2 was added. Here, sk2 does not have ts_recent_stamp set. There are other ordering that will trigger the similar situation below but the idea is the same. 3) When the TCP-ACK-COOKIE comes back, sk2 was selected. "tcp_synq_no_recent_overflow(sk2)" returns true. In this case, all syncookies sent by sk1 will be handled (and rejected) by sk2 while sk1 is still alive. The userspace may create and remove listening SO_REUSEPORT sockets as it sees fit. E.g. Adding new thread (and SO_REUSEPORT sock) to handle incoming requests, old process stopping and new process starting...etc. With or without SO_ATTACH_REUSEPORT_[CB]BPF, the sockets leaving and joining a reuseport group makes picking the same sk to check the syncookie very difficult (if not impossible). The later patches will allow bpf prog more flexibility in deciding where a sk should be located in a bpf map and selecting a particular SO_REUSEPORT sock as it sees fit. e.g. Without closing any sock, replace the whole bpf reuseport_array in one map_update() by using map-in-map. Getting the syncookie check working smoothly across socks in the same "reuse->socks[]" is important. A partial solution is to set the newly added sk's ts_recent_stamp to the max ts_recent_stamp of a reuseport group but that will require to iterate through reuse->socks[] OR pessimistically set it to "now - TCP_SYNCOOKIE_VALID" when a sk is joining a reuseport group. However, neither of them will solve the existing sk getting moved around the reuse->socks[] and that sk may not have ts_recent_stamp updated, unlikely under continuous synflood but not impossible. This patch opts to treat the reuseport group as a whole when considering the last synq overflow timestamp since they are serving the same IP:PORT from the userspace (and BPF program) perspective. "synq_overflow_ts" is added to "struct sock_reuseport". The tcp_synq_overflow() and tcp_synq_no_recent_overflow() will update/check reuse->synq_overflow_ts if the sk is in a reuseport group. Similar to the reuseport decision in __inet_lookup_listener(), both sk->sk_reuseport and sk->sk_reuseport_cb are tested for SO_REUSEPORT usage. Update on "synq_overflow_ts" happens at roughly once every second. A synflood test was done with a 16 rx-queues and 16 reuseport sockets. No meaningful performance change is observed. Before and after the change is ~9Mpps in IPv4. Cc: Eric Dumazet <edumazet@google.com> Signed-off-by: Martin KaFai Lau <kafai@fb.com> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
2018-08-08 16:01:21 +08:00
WRITE_ONCE(reuse->synq_overflow_ts, now);
return;
}
}
last_overflow = READ_ONCE(tcp_sk(sk)->rx_opt.ts_recent_stamp);
tcp: fix rejected syncookies due to stale timestamps If no synflood happens for a long enough period of time, then the synflood timestamp isn't refreshed and jiffies can advance so much that time_after32() can't accurately compare them any more. Therefore, we can end up in a situation where time_after32(now, last_overflow + HZ) returns false, just because these two values are too far apart. In that case, the synflood timestamp isn't updated as it should be, which can trick tcp_synq_no_recent_overflow() into rejecting valid syncookies. For example, let's consider the following scenario on a system with HZ=1000: * The synflood timestamp is 0, either because that's the timestamp of the last synflood or, more commonly, because we're working with a freshly created socket. * We receive a new SYN, which triggers synflood protection. Let's say that this happens when jiffies == 2147484649 (that is, 'synflood timestamp' + HZ + 2^31 + 1). * Then tcp_synq_overflow() doesn't update the synflood timestamp, because time_after32(2147484649, 1000) returns false. With: - 2147484649: the value of jiffies, aka. 'now'. - 1000: the value of 'last_overflow' + HZ. * A bit later, we receive the ACK completing the 3WHS. But cookie_v[46]_check() rejects it because tcp_synq_no_recent_overflow() says that we're not under synflood. That's because time_after32(2147484649, 120000) returns false. With: - 2147484649: the value of jiffies, aka. 'now'. - 120000: the value of 'last_overflow' + TCP_SYNCOOKIE_VALID. Of course, in reality jiffies would have increased a bit, but this condition will last for the next 119 seconds, which is far enough to accommodate for jiffie's growth. Fix this by updating the overflow timestamp whenever jiffies isn't within the [last_overflow, last_overflow + HZ] range. That shouldn't have any performance impact since the update still happens at most once per second. Now we're guaranteed to have fresh timestamps while under synflood, so tcp_synq_no_recent_overflow() can safely use it with time_after32() in such situations. Stale timestamps can still make tcp_synq_no_recent_overflow() return the wrong verdict when not under synflood. This will be handled in the next patch. For 64 bits architectures, the problem was introduced with the conversion of ->tw_ts_recent_stamp to 32 bits integer by commit cca9bab1b72c ("tcp: use monotonic timestamps for PAWS"). The problem has always been there on 32 bits architectures. Fixes: cca9bab1b72c ("tcp: use monotonic timestamps for PAWS") Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Guillaume Nault <gnault@redhat.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2019-12-06 19:38:36 +08:00
if (!time_between32(now, last_overflow, last_overflow + HZ))
WRITE_ONCE(tcp_sk(sk)->rx_opt.ts_recent_stamp, now);
}
/* syncookies: no recent synqueue overflow on this listening socket? */
static inline bool tcp_synq_no_recent_overflow(const struct sock *sk)
{
tcp: Avoid TCP syncookie rejected by SO_REUSEPORT socket Although the actual cookie check "__cookie_v[46]_check()" does not involve sk specific info, it checks whether the sk has recent synq overflow event in "tcp_synq_no_recent_overflow()". The tcp_sk(sk)->rx_opt.ts_recent_stamp is updated every second when it has sent out a syncookie (through "tcp_synq_overflow()"). The above per sk "recent synq overflow event timestamp" works well for non SO_REUSEPORT use case. However, it may cause random connection request reject/discard when SO_REUSEPORT is used with syncookie because it fails the "tcp_synq_no_recent_overflow()" test. When SO_REUSEPORT is used, it usually has multiple listening socks serving TCP connection requests destinated to the same local IP:PORT. There are cases that the TCP-ACK-COOKIE may not be received by the same sk that sent out the syncookie. For example, if reuse->socks[] began with {sk0, sk1}, 1) sk1 sent out syncookies and tcp_sk(sk1)->rx_opt.ts_recent_stamp was updated. 2) the reuse->socks[] became {sk1, sk2} later. e.g. sk0 was first closed and then sk2 was added. Here, sk2 does not have ts_recent_stamp set. There are other ordering that will trigger the similar situation below but the idea is the same. 3) When the TCP-ACK-COOKIE comes back, sk2 was selected. "tcp_synq_no_recent_overflow(sk2)" returns true. In this case, all syncookies sent by sk1 will be handled (and rejected) by sk2 while sk1 is still alive. The userspace may create and remove listening SO_REUSEPORT sockets as it sees fit. E.g. Adding new thread (and SO_REUSEPORT sock) to handle incoming requests, old process stopping and new process starting...etc. With or without SO_ATTACH_REUSEPORT_[CB]BPF, the sockets leaving and joining a reuseport group makes picking the same sk to check the syncookie very difficult (if not impossible). The later patches will allow bpf prog more flexibility in deciding where a sk should be located in a bpf map and selecting a particular SO_REUSEPORT sock as it sees fit. e.g. Without closing any sock, replace the whole bpf reuseport_array in one map_update() by using map-in-map. Getting the syncookie check working smoothly across socks in the same "reuse->socks[]" is important. A partial solution is to set the newly added sk's ts_recent_stamp to the max ts_recent_stamp of a reuseport group but that will require to iterate through reuse->socks[] OR pessimistically set it to "now - TCP_SYNCOOKIE_VALID" when a sk is joining a reuseport group. However, neither of them will solve the existing sk getting moved around the reuse->socks[] and that sk may not have ts_recent_stamp updated, unlikely under continuous synflood but not impossible. This patch opts to treat the reuseport group as a whole when considering the last synq overflow timestamp since they are serving the same IP:PORT from the userspace (and BPF program) perspective. "synq_overflow_ts" is added to "struct sock_reuseport". The tcp_synq_overflow() and tcp_synq_no_recent_overflow() will update/check reuse->synq_overflow_ts if the sk is in a reuseport group. Similar to the reuseport decision in __inet_lookup_listener(), both sk->sk_reuseport and sk->sk_reuseport_cb are tested for SO_REUSEPORT usage. Update on "synq_overflow_ts" happens at roughly once every second. A synflood test was done with a 16 rx-queues and 16 reuseport sockets. No meaningful performance change is observed. Before and after the change is ~9Mpps in IPv4. Cc: Eric Dumazet <edumazet@google.com> Signed-off-by: Martin KaFai Lau <kafai@fb.com> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
2018-08-08 16:01:21 +08:00
unsigned int last_overflow;
unsigned int now = jiffies;
tcp: Avoid TCP syncookie rejected by SO_REUSEPORT socket Although the actual cookie check "__cookie_v[46]_check()" does not involve sk specific info, it checks whether the sk has recent synq overflow event in "tcp_synq_no_recent_overflow()". The tcp_sk(sk)->rx_opt.ts_recent_stamp is updated every second when it has sent out a syncookie (through "tcp_synq_overflow()"). The above per sk "recent synq overflow event timestamp" works well for non SO_REUSEPORT use case. However, it may cause random connection request reject/discard when SO_REUSEPORT is used with syncookie because it fails the "tcp_synq_no_recent_overflow()" test. When SO_REUSEPORT is used, it usually has multiple listening socks serving TCP connection requests destinated to the same local IP:PORT. There are cases that the TCP-ACK-COOKIE may not be received by the same sk that sent out the syncookie. For example, if reuse->socks[] began with {sk0, sk1}, 1) sk1 sent out syncookies and tcp_sk(sk1)->rx_opt.ts_recent_stamp was updated. 2) the reuse->socks[] became {sk1, sk2} later. e.g. sk0 was first closed and then sk2 was added. Here, sk2 does not have ts_recent_stamp set. There are other ordering that will trigger the similar situation below but the idea is the same. 3) When the TCP-ACK-COOKIE comes back, sk2 was selected. "tcp_synq_no_recent_overflow(sk2)" returns true. In this case, all syncookies sent by sk1 will be handled (and rejected) by sk2 while sk1 is still alive. The userspace may create and remove listening SO_REUSEPORT sockets as it sees fit. E.g. Adding new thread (and SO_REUSEPORT sock) to handle incoming requests, old process stopping and new process starting...etc. With or without SO_ATTACH_REUSEPORT_[CB]BPF, the sockets leaving and joining a reuseport group makes picking the same sk to check the syncookie very difficult (if not impossible). The later patches will allow bpf prog more flexibility in deciding where a sk should be located in a bpf map and selecting a particular SO_REUSEPORT sock as it sees fit. e.g. Without closing any sock, replace the whole bpf reuseport_array in one map_update() by using map-in-map. Getting the syncookie check working smoothly across socks in the same "reuse->socks[]" is important. A partial solution is to set the newly added sk's ts_recent_stamp to the max ts_recent_stamp of a reuseport group but that will require to iterate through reuse->socks[] OR pessimistically set it to "now - TCP_SYNCOOKIE_VALID" when a sk is joining a reuseport group. However, neither of them will solve the existing sk getting moved around the reuse->socks[] and that sk may not have ts_recent_stamp updated, unlikely under continuous synflood but not impossible. This patch opts to treat the reuseport group as a whole when considering the last synq overflow timestamp since they are serving the same IP:PORT from the userspace (and BPF program) perspective. "synq_overflow_ts" is added to "struct sock_reuseport". The tcp_synq_overflow() and tcp_synq_no_recent_overflow() will update/check reuse->synq_overflow_ts if the sk is in a reuseport group. Similar to the reuseport decision in __inet_lookup_listener(), both sk->sk_reuseport and sk->sk_reuseport_cb are tested for SO_REUSEPORT usage. Update on "synq_overflow_ts" happens at roughly once every second. A synflood test was done with a 16 rx-queues and 16 reuseport sockets. No meaningful performance change is observed. Before and after the change is ~9Mpps in IPv4. Cc: Eric Dumazet <edumazet@google.com> Signed-off-by: Martin KaFai Lau <kafai@fb.com> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
2018-08-08 16:01:21 +08:00
if (sk->sk_reuseport) {
struct sock_reuseport *reuse;
reuse = rcu_dereference(sk->sk_reuseport_cb);
if (likely(reuse)) {
last_overflow = READ_ONCE(reuse->synq_overflow_ts);
tcp: tighten acceptance of ACKs not matching a child socket When no synflood occurs, the synflood timestamp isn't updated. Therefore it can be so old that time_after32() can consider it to be in the future. That's a problem for tcp_synq_no_recent_overflow() as it may report that a recent overflow occurred while, in fact, it's just that jiffies has grown past 'last_overflow' + TCP_SYNCOOKIE_VALID + 2^31. Spurious detection of recent overflows lead to extra syncookie verification in cookie_v[46]_check(). At that point, the verification should fail and the packet dropped. But we should have dropped the packet earlier as we didn't even send a syncookie. Let's refine tcp_synq_no_recent_overflow() to report a recent overflow only if jiffies is within the [last_overflow, last_overflow + TCP_SYNCOOKIE_VALID] interval. This way, no spurious recent overflow is reported when jiffies wraps and 'last_overflow' becomes in the future from the point of view of time_after32(). However, if jiffies wraps and enters the [last_overflow, last_overflow + TCP_SYNCOOKIE_VALID] interval (with 'last_overflow' being a stale synflood timestamp), then tcp_synq_no_recent_overflow() still erroneously reports an overflow. In such cases, we have to rely on syncookie verification to drop the packet. We unfortunately have no way to differentiate between a fresh and a stale syncookie timestamp. In practice, using last_overflow as lower bound is problematic. If the synflood timestamp is concurrently updated between the time we read jiffies and the moment we store the timestamp in 'last_overflow', then 'now' becomes smaller than 'last_overflow' and tcp_synq_no_recent_overflow() returns true, potentially dropping a valid syncookie. Reading jiffies after loading the timestamp could fix the problem, but that'd require a memory barrier. Let's just accommodate for potential timestamp growth instead and extend the interval using 'last_overflow - HZ' as lower bound. Signed-off-by: Guillaume Nault <gnault@redhat.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2019-12-06 19:38:43 +08:00
return !time_between32(now, last_overflow - HZ,
last_overflow +
TCP_SYNCOOKIE_VALID);
tcp: Avoid TCP syncookie rejected by SO_REUSEPORT socket Although the actual cookie check "__cookie_v[46]_check()" does not involve sk specific info, it checks whether the sk has recent synq overflow event in "tcp_synq_no_recent_overflow()". The tcp_sk(sk)->rx_opt.ts_recent_stamp is updated every second when it has sent out a syncookie (through "tcp_synq_overflow()"). The above per sk "recent synq overflow event timestamp" works well for non SO_REUSEPORT use case. However, it may cause random connection request reject/discard when SO_REUSEPORT is used with syncookie because it fails the "tcp_synq_no_recent_overflow()" test. When SO_REUSEPORT is used, it usually has multiple listening socks serving TCP connection requests destinated to the same local IP:PORT. There are cases that the TCP-ACK-COOKIE may not be received by the same sk that sent out the syncookie. For example, if reuse->socks[] began with {sk0, sk1}, 1) sk1 sent out syncookies and tcp_sk(sk1)->rx_opt.ts_recent_stamp was updated. 2) the reuse->socks[] became {sk1, sk2} later. e.g. sk0 was first closed and then sk2 was added. Here, sk2 does not have ts_recent_stamp set. There are other ordering that will trigger the similar situation below but the idea is the same. 3) When the TCP-ACK-COOKIE comes back, sk2 was selected. "tcp_synq_no_recent_overflow(sk2)" returns true. In this case, all syncookies sent by sk1 will be handled (and rejected) by sk2 while sk1 is still alive. The userspace may create and remove listening SO_REUSEPORT sockets as it sees fit. E.g. Adding new thread (and SO_REUSEPORT sock) to handle incoming requests, old process stopping and new process starting...etc. With or without SO_ATTACH_REUSEPORT_[CB]BPF, the sockets leaving and joining a reuseport group makes picking the same sk to check the syncookie very difficult (if not impossible). The later patches will allow bpf prog more flexibility in deciding where a sk should be located in a bpf map and selecting a particular SO_REUSEPORT sock as it sees fit. e.g. Without closing any sock, replace the whole bpf reuseport_array in one map_update() by using map-in-map. Getting the syncookie check working smoothly across socks in the same "reuse->socks[]" is important. A partial solution is to set the newly added sk's ts_recent_stamp to the max ts_recent_stamp of a reuseport group but that will require to iterate through reuse->socks[] OR pessimistically set it to "now - TCP_SYNCOOKIE_VALID" when a sk is joining a reuseport group. However, neither of them will solve the existing sk getting moved around the reuse->socks[] and that sk may not have ts_recent_stamp updated, unlikely under continuous synflood but not impossible. This patch opts to treat the reuseport group as a whole when considering the last synq overflow timestamp since they are serving the same IP:PORT from the userspace (and BPF program) perspective. "synq_overflow_ts" is added to "struct sock_reuseport". The tcp_synq_overflow() and tcp_synq_no_recent_overflow() will update/check reuse->synq_overflow_ts if the sk is in a reuseport group. Similar to the reuseport decision in __inet_lookup_listener(), both sk->sk_reuseport and sk->sk_reuseport_cb are tested for SO_REUSEPORT usage. Update on "synq_overflow_ts" happens at roughly once every second. A synflood test was done with a 16 rx-queues and 16 reuseport sockets. No meaningful performance change is observed. Before and after the change is ~9Mpps in IPv4. Cc: Eric Dumazet <edumazet@google.com> Signed-off-by: Martin KaFai Lau <kafai@fb.com> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
2018-08-08 16:01:21 +08:00
}
}
last_overflow = READ_ONCE(tcp_sk(sk)->rx_opt.ts_recent_stamp);
tcp: tighten acceptance of ACKs not matching a child socket When no synflood occurs, the synflood timestamp isn't updated. Therefore it can be so old that time_after32() can consider it to be in the future. That's a problem for tcp_synq_no_recent_overflow() as it may report that a recent overflow occurred while, in fact, it's just that jiffies has grown past 'last_overflow' + TCP_SYNCOOKIE_VALID + 2^31. Spurious detection of recent overflows lead to extra syncookie verification in cookie_v[46]_check(). At that point, the verification should fail and the packet dropped. But we should have dropped the packet earlier as we didn't even send a syncookie. Let's refine tcp_synq_no_recent_overflow() to report a recent overflow only if jiffies is within the [last_overflow, last_overflow + TCP_SYNCOOKIE_VALID] interval. This way, no spurious recent overflow is reported when jiffies wraps and 'last_overflow' becomes in the future from the point of view of time_after32(). However, if jiffies wraps and enters the [last_overflow, last_overflow + TCP_SYNCOOKIE_VALID] interval (with 'last_overflow' being a stale synflood timestamp), then tcp_synq_no_recent_overflow() still erroneously reports an overflow. In such cases, we have to rely on syncookie verification to drop the packet. We unfortunately have no way to differentiate between a fresh and a stale syncookie timestamp. In practice, using last_overflow as lower bound is problematic. If the synflood timestamp is concurrently updated between the time we read jiffies and the moment we store the timestamp in 'last_overflow', then 'now' becomes smaller than 'last_overflow' and tcp_synq_no_recent_overflow() returns true, potentially dropping a valid syncookie. Reading jiffies after loading the timestamp could fix the problem, but that'd require a memory barrier. Let's just accommodate for potential timestamp growth instead and extend the interval using 'last_overflow - HZ' as lower bound. Signed-off-by: Guillaume Nault <gnault@redhat.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2019-12-06 19:38:43 +08:00
/* If last_overflow <= jiffies <= last_overflow + TCP_SYNCOOKIE_VALID,
* then we're under synflood. However, we have to use
* 'last_overflow - HZ' as lower bound. That's because a concurrent
* tcp_synq_overflow() could update .ts_recent_stamp after we read
* jiffies but before we store .ts_recent_stamp into last_overflow,
* which could lead to rejecting a valid syncookie.
*/
return !time_between32(now, last_overflow - HZ,
last_overflow + TCP_SYNCOOKIE_VALID);
}
static inline u32 tcp_cookie_time(void)
{
u64 val = get_jiffies_64();
do_div(val, TCP_SYNCOOKIE_PERIOD);
return val;
}
u32 __cookie_v4_init_sequence(const struct iphdr *iph, const struct tcphdr *th,
u16 *mssp);
__u32 cookie_v4_init_sequence(const struct sk_buff *skb, __u16 *mss);
u64 cookie_init_timestamp(struct request_sock *req, u64 now);
bool cookie_timestamp_decode(const struct net *net,
struct tcp_options_received *opt);
syncookies: split cookie_check_timestamp() into two functions The function cookie_check_timestamp(), both called from IPv4/6 context, is being used to decode the echoed timestamp from the SYN/ACK into TCP options used for follow-up communication with the peer. We can remove ECN handling from that function, split it into a separate one, and simply rename the original function into cookie_decode_options(). cookie_decode_options() just fills in tcp_option struct based on the echoed timestamp received from the peer. Anything that fails in this function will actually discard the request socket. While this is the natural place for decoding options such as ECN which commit 172d69e63c7f ("syncookies: add support for ECN") added, we argue that in particular for ECN handling, it can be checked at a later point in time as the request sock would actually not need to be dropped from this, but just ECN support turned off. Therefore, we split this functionality into cookie_ecn_ok(), which tells us if the timestamp indicates ECN support AND the tcp_ecn sysctl is enabled. This prepares for per-route ECN support: just looking at the tcp_ecn sysctl won't be enough anymore at that point; if the timestamp indicates ECN and sysctl tcp_ecn == 0, we will also need to check the ECN dst metric. This would mean adding a route lookup to cookie_check_timestamp(), which we definitely want to avoid. As we already do a route lookup at a later point in cookie_{v4,v6}_check(), we can simply make use of that as well for the new cookie_ecn_ok() function w/o any additional cost. Joint work with Daniel Borkmann. Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Daniel Borkmann <dborkman@redhat.com> Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-11-04 00:35:02 +08:00
bool cookie_ecn_ok(const struct tcp_options_received *opt,
net: allow setting ecn via routing table This patch allows to set ECN on a per-route basis in case the sysctl tcp_ecn is not set to 1. In other words, when ECN is set for specific routes, it provides a tcp_ecn=1 behaviour for that route while the rest of the stack acts according to the global settings. One can use 'ip route change dev $dev $net features ecn' to toggle this. Having a more fine-grained per-route setting can be beneficial for various reasons, for example, 1) within data centers, or 2) local ISPs may deploy ECN support for their own video/streaming services [1], etc. There was a recent measurement study/paper [2] which scanned the Alexa's publicly available top million websites list from a vantage point in US, Europe and Asia: Half of the Alexa list will now happily use ECN (tcp_ecn=2, most likely blamed to commit 255cac91c3 ("tcp: extend ECN sysctl to allow server-side only ECN") ;)); the break in connectivity on-path was found is about 1 in 10,000 cases. Timeouts rather than receiving back RSTs were much more common in the negotiation phase (and mostly seen in the Alexa middle band, ranks around 50k-150k): from 12-thousand hosts on which there _may_ be ECN-linked connection failures, only 79 failed with RST when _not_ failing with RST when ECN is not requested. It's unclear though, how much equipment in the wild actually marks CE when buffers start to fill up. We thought about a fallback to non-ECN for retransmitted SYNs as another global option (which could perhaps one day be made default), but as Eric points out, there's much more work needed to detect broken middleboxes. Two examples Eric mentioned are buggy firewalls that accept only a single SYN per flow, and middleboxes that successfully let an ECN flow establish, but later mark CE for all packets (so cwnd converges to 1). [1] http://www.ietf.org/proceedings/89/slides/slides-89-tsvarea-1.pdf, p.15 [2] http://ecn.ethz.ch/ Joint work with Daniel Borkmann. Reference: http://thread.gmane.org/gmane.linux.network/335797 Suggested-by: Hannes Frederic Sowa <hannes@stressinduktion.org> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Daniel Borkmann <dborkman@redhat.com> Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-11-04 00:35:03 +08:00
const struct net *net, const struct dst_entry *dst);
/* From net/ipv6/syncookies.c */
int __cookie_v6_check(const struct ipv6hdr *iph, const struct tcphdr *th,
u32 cookie);
struct sock *cookie_v6_check(struct sock *sk, struct sk_buff *skb);
syncookies: split cookie_check_timestamp() into two functions The function cookie_check_timestamp(), both called from IPv4/6 context, is being used to decode the echoed timestamp from the SYN/ACK into TCP options used for follow-up communication with the peer. We can remove ECN handling from that function, split it into a separate one, and simply rename the original function into cookie_decode_options(). cookie_decode_options() just fills in tcp_option struct based on the echoed timestamp received from the peer. Anything that fails in this function will actually discard the request socket. While this is the natural place for decoding options such as ECN which commit 172d69e63c7f ("syncookies: add support for ECN") added, we argue that in particular for ECN handling, it can be checked at a later point in time as the request sock would actually not need to be dropped from this, but just ECN support turned off. Therefore, we split this functionality into cookie_ecn_ok(), which tells us if the timestamp indicates ECN support AND the tcp_ecn sysctl is enabled. This prepares for per-route ECN support: just looking at the tcp_ecn sysctl won't be enough anymore at that point; if the timestamp indicates ECN and sysctl tcp_ecn == 0, we will also need to check the ECN dst metric. This would mean adding a route lookup to cookie_check_timestamp(), which we definitely want to avoid. As we already do a route lookup at a later point in cookie_{v4,v6}_check(), we can simply make use of that as well for the new cookie_ecn_ok() function w/o any additional cost. Joint work with Daniel Borkmann. Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Daniel Borkmann <dborkman@redhat.com> Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-11-04 00:35:02 +08:00
u32 __cookie_v6_init_sequence(const struct ipv6hdr *iph,
const struct tcphdr *th, u16 *mssp);
__u32 cookie_v6_init_sequence(const struct sk_buff *skb, __u16 *mss);
#endif
/* tcp_output.c */
void __tcp_push_pending_frames(struct sock *sk, unsigned int cur_mss,
int nonagle);
int __tcp_retransmit_skb(struct sock *sk, struct sk_buff *skb, int segs);
int tcp_retransmit_skb(struct sock *sk, struct sk_buff *skb, int segs);
void tcp_retransmit_timer(struct sock *sk);
void tcp_xmit_retransmit_queue(struct sock *);
void tcp_simple_retransmit(struct sock *);
tcp: add reordering timer in RACK loss detection This patch makes RACK install a reordering timer when it suspects some packets might be lost, but wants to delay the decision a little bit to accomodate reordering. It does not create a new timer but instead repurposes the existing RTO timer, because both are meant to retransmit packets. Specifically it arms a timer ICSK_TIME_REO_TIMEOUT when the RACK timing check fails. The wait time is set to RACK.RTT + RACK.reo_wnd - (NOW - Packet.xmit_time) + fudge This translates to expecting a packet (Packet) should take (RACK.RTT + RACK.reo_wnd + fudge) to deliver after it was sent. When there are multiple packets that need a timer, we use one timer with the maximum timeout. Therefore the timer conservatively uses the maximum window to expire N packets by one timeout, instead of N timeouts to expire N packets sent at different times. The fudge factor is 2 jiffies to ensure when the timer fires, all the suspected packets would exceed the deadline and be marked lost by tcp_rack_detect_loss(). It has to be at least 1 jiffy because the clock may tick between calling icsk_reset_xmit_timer(timeout) and actually hang the timer. The next jiffy is to lower-bound the timeout to 2 jiffies when reo_wnd is < 1ms. When the reordering timer fires (tcp_rack_reo_timeout): If we aren't in Recovery we'll enter fast recovery and force fast retransmit. This is very similar to the early retransmit (RFC5827) except RACK is not constrained to only enter recovery for small outstanding flights. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-01-13 14:11:33 +08:00
void tcp_enter_recovery(struct sock *sk, bool ece_ack);
int tcp_trim_head(struct sock *, struct sk_buff *, u32);
tcp: implement rb-tree based retransmit queue Using a linear list to store all skbs in write queue has been okay for quite a while : O(N) is not too bad when N < 500. Things get messy when N is the order of 100,000 : Modern TCP stacks want 10Gbit+ of throughput even with 200 ms RTT flows. 40 ns per cache line miss means a full scan can use 4 ms, blowing away CPU caches. SACK processing often can use various hints to avoid parsing whole retransmit queue. But with high packet losses and/or high reordering, hints no longer work. Sender has to process thousands of unfriendly SACK, accumulating a huge socket backlog, burning a cpu and massively dropping packets. Using an rb-tree for retransmit queue has been avoided for years because it added complexity and overhead, but now is the time to be more resistant and say no to quadratic behavior. 1) RTX queue is no longer part of the write queue : already sent skbs are stored in one rb-tree. 2) Since reaching the head of write queue no longer needs sk->sk_send_head, we added an union of sk_send_head and tcp_rtx_queue Tested: On receiver : netem on ingress : delay 150ms 200us loss 1 GRO disabled to force stress and SACK storms. for f in `seq 1 10` do ./netperf -H lpaa6 -l30 -- -K bbr -o THROUGHPUT|tail -1 done | awk '{print $0} {sum += $0} END {printf "%7u\n",sum}' Before patch : 323.87 351.48 339.59 338.62 306.72 204.07 304.93 291.88 202.47 176.88 2840 After patch: 1700.83 2207.98 2070.17 1544.26 2114.76 2124.89 1693.14 1080.91 2216.82 1299.94 18053 Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-10-06 13:21:27 +08:00
enum tcp_queue {
TCP_FRAG_IN_WRITE_QUEUE,
TCP_FRAG_IN_RTX_QUEUE,
};
int tcp_fragment(struct sock *sk, enum tcp_queue tcp_queue,
struct sk_buff *skb, u32 len,
unsigned int mss_now, gfp_t gfp);
void tcp_send_probe0(struct sock *);
void tcp_send_partial(struct sock *);
int tcp_write_wakeup(struct sock *, int mib);
void tcp_send_fin(struct sock *sk);
void tcp_send_active_reset(struct sock *sk, gfp_t priority);
int tcp_send_synack(struct sock *);
void tcp_push_one(struct sock *, unsigned int mss_now);
void __tcp_send_ack(struct sock *sk, u32 rcv_nxt);
void tcp_send_ack(struct sock *sk);
void tcp_send_delayed_ack(struct sock *sk);
void tcp_send_loss_probe(struct sock *sk);
tcp: when scheduling TLP, time of RTO should account for current ACK Fix the TLP scheduling logic so that when scheduling a TLP probe, we ensure that the estimated time at which an RTO would fire accounts for the fact that ACKs indicating forward progress should push back RTO times. After the following fix: df92c8394e6e ("tcp: fix xmit timer to only be reset if data ACKed/SACKed") we had an unintentional behavior change in the following kind of scenario: suppose the RTT variance has been very low recently. Then suppose we send out a flight of N packets and our RTT is 100ms: t=0: send a flight of N packets t=100ms: receive an ACK for N-1 packets The response before df92c8394e6e that was: -> schedule a TLP for now + RTO_interval The response after df92c8394e6e is: -> schedule a TLP for t=0 + RTO_interval Since RTO_interval = srtt + RTT_variance, this means that we have scheduled a TLP timer at a point in the future that only accounts for RTT_variance. If the RTT_variance term is small, this means that the timer fires soon. Before df92c8394e6e this would not happen, because in that code, when we receive an ACK for a prefix of flight, we did: 1) Near the top of tcp_ack(), switch from TLP timer to RTO at write_queue_head->paket_tx_time + RTO_interval: if (icsk->icsk_pending == ICSK_TIME_LOSS_PROBE) tcp_rearm_rto(sk); 2) In tcp_clean_rtx_queue(), update the RTO to now + RTO_interval: if (flag & FLAG_ACKED) { tcp_rearm_rto(sk); 3) In tcp_ack() after tcp_fastretrans_alert() switch from RTO to TLP at now + RTO_interval: if (icsk->icsk_pending == ICSK_TIME_RETRANS) tcp_schedule_loss_probe(sk); In df92c8394e6e we removed that 3-phase dance, and instead directly set the TLP timer once: we set the TLP timer in cases like this to write_queue_head->packet_tx_time + RTO_interval. So if the RTT variance is small, then this means that this is setting the TLP timer to fire quite soon. This means if the ACK for the tail of the flight takes longer than an RTT to arrive (often due to delayed ACKs), then the TLP timer fires too quickly. Fixes: df92c8394e6e ("tcp: fix xmit timer to only be reset if data ACKed/SACKed") Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-11-18 10:06:14 +08:00
bool tcp_schedule_loss_probe(struct sock *sk, bool advancing_rto);
tcp: Merge tx_flags and tskey in tcp_shifted_skb After receiving sacks, tcp_shifted_skb() will collapse skbs if possible. tx_flags and tskey also have to be merged. This patch reuses the tcp_skb_collapse_tstamp() to handle them. BPF Output Before: ~~~~~ <no-output-due-to-missing-tstamp-event> BPF Output After: ~~~~~ <...>-2024 [007] d.s. 88.644374: : ee_data:14599 Packetdrill Script: ~~~~~ +0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10` +0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1` +0 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7> 0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7> 0.200 < . 1:1(0) ack 1 win 257 0.200 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0 0.200 write(4, ..., 1460) = 1460 +0 setsockopt(4, SOL_SOCKET, 37, [2688], 4) = 0 0.200 write(4, ..., 13140) = 13140 0.200 > P. 1:1461(1460) ack 1 0.200 > . 1461:8761(7300) ack 1 0.200 > P. 8761:14601(5840) ack 1 0.300 < . 1:1(0) ack 1 win 257 <sack 1461:14601,nop,nop> 0.300 > P. 1:1461(1460) ack 1 0.400 < . 1:1(0) ack 14601 win 257 0.400 close(4) = 0 0.400 > F. 14601:14601(0) ack 1 0.500 < F. 1:1(0) ack 14602 win 257 0.500 > . 14602:14602(0) ack 2 Signed-off-by: Martin KaFai Lau <kafai@fb.com> Cc: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Soheil Hassas Yeganeh <soheil@google.com> Cc: Willem de Bruijn <willemb@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Tested-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-04-20 13:39:29 +08:00
void tcp_skb_collapse_tstamp(struct sk_buff *skb,
const struct sk_buff *next_skb);
/* tcp_input.c */
void tcp_rearm_rto(struct sock *sk);
tcp: usec resolution SYN/ACK RTT Currently SYN/ACK RTT is measured in jiffies. For LAN the SYN/ACK RTT is often measured as 0ms or sometimes 1ms, which would affect RTT estimation and min RTT samping used by some congestion control. This patch improves SYN/ACK RTT to be usec resolution if platform supports it. While the timestamping of SYN/ACK is done in request sock, the RTT measurement is carefully arranged to avoid storing another u64 timestamp in tcp_sock. For regular handshake w/o SYNACK retransmission, the RTT is sampled right after the child socket is created and right before the request sock is released (tcp_check_req() in tcp_minisocks.c) For Fast Open the child socket is already created when SYN/ACK was sent, the RTT is sampled in tcp_rcv_state_process() after processing the final ACK an right before the request socket is released. If the SYN/ACK was retransmistted or SYN-cookie was used, we rely on TCP timestamps to measure the RTT. The sample is taken at the same place in tcp_rcv_state_process() after the timestamp values are validated in tcp_validate_incoming(). Note that we do not store TS echo value in request_sock for SYN-cookies, because the value is already stored in tp->rx_opt used by tcp_ack_update_rtt(). One side benefit is that the RTT measurement now happens before initializing congestion control (of the passive side). Therefore the congestion control can use the SYN/ACK RTT. 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>
2015-09-19 02:36:14 +08:00
void tcp_synack_rtt_meas(struct sock *sk, struct request_sock *req);
void tcp_reset(struct sock *sk);
void tcp_skb_mark_lost_uncond_verify(struct tcp_sock *tp, struct sk_buff *skb);
void tcp_fin(struct sock *sk);
/* tcp_timer.c */
void tcp_init_xmit_timers(struct sock *);
static inline void tcp_clear_xmit_timers(struct sock *sk)
{
if (hrtimer_try_to_cancel(&tcp_sk(sk)->pacing_timer) == 1)
__sock_put(sk);
if (hrtimer_try_to_cancel(&tcp_sk(sk)->compressed_ack_timer) == 1)
__sock_put(sk);
inet_csk_clear_xmit_timers(sk);
}
unsigned int tcp_sync_mss(struct sock *sk, u32 pmtu);
unsigned int tcp_current_mss(struct sock *sk);
/* Bound MSS / TSO packet size with the half of the window */
static inline int tcp_bound_to_half_wnd(struct tcp_sock *tp, int pktsize)
{
int cutoff;
/* When peer uses tiny windows, there is no use in packetizing
* to sub-MSS pieces for the sake of SWS or making sure there
* are enough packets in the pipe for fast recovery.
*
* On the other hand, for extremely large MSS devices, handling
* smaller than MSS windows in this way does make sense.
*/
if (tp->max_window > TCP_MSS_DEFAULT)
cutoff = (tp->max_window >> 1);
else
cutoff = tp->max_window;
if (cutoff && pktsize > cutoff)
return max_t(int, cutoff, 68U - tp->tcp_header_len);
else
return pktsize;
}
/* tcp.c */
void tcp_get_info(struct sock *, struct tcp_info *);
/* Read 'sendfile()'-style from a TCP socket */
int tcp_read_sock(struct sock *sk, read_descriptor_t *desc,
sk_read_actor_t recv_actor);
void tcp_initialize_rcv_mss(struct sock *sk);
int tcp_mtu_to_mss(struct sock *sk, int pmtu);
int tcp_mss_to_mtu(struct sock *sk, int mss);
void tcp_mtup_init(struct sock *sk);
static inline void tcp_bound_rto(const struct sock *sk)
{
if (inet_csk(sk)->icsk_rto > TCP_RTO_MAX)
inet_csk(sk)->icsk_rto = TCP_RTO_MAX;
}
static inline u32 __tcp_set_rto(const struct tcp_sock *tp)
{
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 06:02:48 +08:00
return usecs_to_jiffies((tp->srtt_us >> 3) + tp->rttvar_us);
}
static inline void __tcp_fast_path_on(struct tcp_sock *tp, u32 snd_wnd)
{
tp->pred_flags = htonl((tp->tcp_header_len << 26) |
ntohl(TCP_FLAG_ACK) |
snd_wnd);
}
static inline void tcp_fast_path_on(struct tcp_sock *tp)
{
__tcp_fast_path_on(tp, tp->snd_wnd >> tp->rx_opt.snd_wscale);
}
static inline void tcp_fast_path_check(struct sock *sk)
{
struct tcp_sock *tp = tcp_sk(sk);
if (RB_EMPTY_ROOT(&tp->out_of_order_queue) &&
tp->rcv_wnd &&
atomic_read(&sk->sk_rmem_alloc) < sk->sk_rcvbuf &&
!tp->urg_data)
tcp_fast_path_on(tp);
}
/* Compute the actual rto_min value */
static inline u32 tcp_rto_min(struct sock *sk)
{
const struct dst_entry *dst = __sk_dst_get(sk);
u32 rto_min = inet_csk(sk)->icsk_rto_min;
if (dst && dst_metric_locked(dst, RTAX_RTO_MIN))
rto_min = dst_metric_rtt(dst, RTAX_RTO_MIN);
return rto_min;
}
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 06:02:48 +08:00
static inline u32 tcp_rto_min_us(struct sock *sk)
{
return jiffies_to_usecs(tcp_rto_min(sk));
}
net: tcp: add per route congestion control This work adds the possibility to define a per route/destination congestion control algorithm. Generally, this opens up the possibility for a machine with different links to enforce specific congestion control algorithms with optimal strategies for each of them based on their network characteristics, even transparently for a single application listening on all links. For our specific use case, this additionally facilitates deployment of DCTCP, for example, applications can easily serve internal traffic/dsts in DCTCP and external one with CUBIC. Other scenarios would also allow for utilizing e.g. long living, low priority background flows for certain destinations/routes while still being able for normal traffic to utilize the default congestion control algorithm. We also thought about a per netns setting (where different defaults are possible), but given its actually a link specific property, we argue that a per route/destination setting is the most natural and flexible. The administrator can utilize this through ip-route(8) by appending "congctl [lock] <name>", where <name> denotes the name of a congestion control algorithm and the optional lock parameter allows to enforce the given algorithm so that applications in user space would not be allowed to overwrite that algorithm for that destination. The dst metric lookups are being done when a dst entry is already available in order to avoid a costly lookup and still before the algorithms are being initialized, thus overhead is very low when the feature is not being used. While the client side would need to drop the current reference on the module, on server side this can actually even be avoided as we just got a flat-copied socket clone. Joint work with Florian Westphal. Suggested-by: Hannes Frederic Sowa <hannes@stressinduktion.org> Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: Daniel Borkmann <dborkman@redhat.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:48 +08:00
static inline bool tcp_ca_dst_locked(const struct dst_entry *dst)
{
return dst_metric_locked(dst, RTAX_CC_ALGO);
}
tcp: track min RTT using windowed min-filter Kathleen Nichols' algorithm for tracking the minimum RTT of a data stream over some measurement window. It uses constant space and constant time per update. Yet it almost always delivers the same minimum as an implementation that has to keep all the data in the window. The measurement window is tunable via sysctl.net.ipv4.tcp_min_rtt_wlen with a default value of 5 minutes. The algorithm keeps track of the best, 2nd best & 3rd best min values, maintaining an invariant that the measurement time of the n'th best >= n-1'th best. It also makes sure that the three values are widely separated in the time window since that bounds the worse case error when that data is monotonically increasing over the window. Upon getting a new min, we can forget everything earlier because it has no value - the new min is less than everything else in the window by definition and it's the most recent. So we restart fresh on every new min and overwrites the 2nd & 3rd choices. The same property holds for the 2nd & 3rd best. Therefore we have to maintain two invariants to maximize the information in the samples, one on values (1st.v <= 2nd.v <= 3rd.v) and the other on times (now-win <=1st.t <= 2nd.t <= 3rd.t <= now). These invariants determine the structure of the code The RTT input to the windowed filter is the minimum RTT measured from ACK or SACK, or as the last resort from TCP timestamps. The accessor tcp_min_rtt() returns the minimum RTT seen in the window. ~0U indicates it is not available. The minimum is 1usec even if the true RTT is below that. 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 12:57:42 +08:00
/* Minimum RTT in usec. ~0 means not available. */
static inline u32 tcp_min_rtt(const struct tcp_sock *tp)
{
return minmax_get(&tp->rtt_min);
tcp: track min RTT using windowed min-filter Kathleen Nichols' algorithm for tracking the minimum RTT of a data stream over some measurement window. It uses constant space and constant time per update. Yet it almost always delivers the same minimum as an implementation that has to keep all the data in the window. The measurement window is tunable via sysctl.net.ipv4.tcp_min_rtt_wlen with a default value of 5 minutes. The algorithm keeps track of the best, 2nd best & 3rd best min values, maintaining an invariant that the measurement time of the n'th best >= n-1'th best. It also makes sure that the three values are widely separated in the time window since that bounds the worse case error when that data is monotonically increasing over the window. Upon getting a new min, we can forget everything earlier because it has no value - the new min is less than everything else in the window by definition and it's the most recent. So we restart fresh on every new min and overwrites the 2nd & 3rd choices. The same property holds for the 2nd & 3rd best. Therefore we have to maintain two invariants to maximize the information in the samples, one on values (1st.v <= 2nd.v <= 3rd.v) and the other on times (now-win <=1st.t <= 2nd.t <= 3rd.t <= now). These invariants determine the structure of the code The RTT input to the windowed filter is the minimum RTT measured from ACK or SACK, or as the last resort from TCP timestamps. The accessor tcp_min_rtt() returns the minimum RTT seen in the window. ~0U indicates it is not available. The minimum is 1usec even if the true RTT is below that. 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 12:57:42 +08:00
}
/* Compute the actual receive window we are currently advertising.
* Rcv_nxt can be after the window if our peer push more data
* than the offered window.
*/
static inline u32 tcp_receive_window(const struct tcp_sock *tp)
{
s32 win = tp->rcv_wup + tp->rcv_wnd - tp->rcv_nxt;
if (win < 0)
win = 0;
return (u32) win;
}
/* Choose a new window, without checks for shrinking, and without
* scaling applied to the result. The caller does these things
* if necessary. This is a "raw" window selection.
*/
u32 __tcp_select_window(struct sock *sk);
void tcp_send_window_probe(struct sock *sk);
/* TCP uses 32bit jiffies to save some space.
* Note that this is different from tcp_time_stamp, which
* historically has been the same until linux-4.13.
*/
#define tcp_jiffies32 ((u32)jiffies)
/*
* Deliver a 32bit value for TCP timestamp option (RFC 7323)
* It is no longer tied to jiffies, but to 1 ms clock.
* Note: double check if you want to use tcp_jiffies32 instead of this.
*/
#define TCP_TS_HZ 1000
static inline u64 tcp_clock_ns(void)
{
return ktime_get_ns();
}
static inline u64 tcp_clock_us(void)
{
return div_u64(tcp_clock_ns(), NSEC_PER_USEC);
}
/* This should only be used in contexts where tp->tcp_mstamp is up to date */
static inline u32 tcp_time_stamp(const struct tcp_sock *tp)
{
return div_u64(tp->tcp_mstamp, USEC_PER_SEC / TCP_TS_HZ);
}
/* Convert a nsec timestamp into TCP TSval timestamp (ms based currently) */
static inline u32 tcp_ns_to_ts(u64 ns)
{
return div_u64(ns, NSEC_PER_SEC / TCP_TS_HZ);
}
/* Could use tcp_clock_us() / 1000, but this version uses a single divide */
static inline u32 tcp_time_stamp_raw(void)
{
return tcp_ns_to_ts(tcp_clock_ns());
}
void tcp_mstamp_refresh(struct tcp_sock *tp);
static inline u32 tcp_stamp_us_delta(u64 t1, u64 t0)
{
return max_t(s64, t1 - t0, 0);
}
static inline u32 tcp_skb_timestamp(const struct sk_buff *skb)
{
return tcp_ns_to_ts(skb->skb_mstamp_ns);
}
/* provide the departure time in us unit */
static inline u64 tcp_skb_timestamp_us(const struct sk_buff *skb)
{
return div_u64(skb->skb_mstamp_ns, NSEC_PER_USEC);
}
#define tcp_flag_byte(th) (((u_int8_t *)th)[13])
#define TCPHDR_FIN 0x01
#define TCPHDR_SYN 0x02
#define TCPHDR_RST 0x04
#define TCPHDR_PSH 0x08
#define TCPHDR_ACK 0x10
#define TCPHDR_URG 0x20
#define TCPHDR_ECE 0x40
#define TCPHDR_CWR 0x80
tcp: add rfc3168, section 6.1.1.1. fallback This work as a follow-up of commit f7b3bec6f516 ("net: allow setting ecn via routing table") and adds RFC3168 section 6.1.1.1. fallback for outgoing ECN connections. In other words, this work adds a retry with a non-ECN setup SYN packet, as suggested from the RFC on the first timeout: [...] A host that receives no reply to an ECN-setup SYN within the normal SYN retransmission timeout interval MAY resend the SYN and any subsequent SYN retransmissions with CWR and ECE cleared. [...] Schematic client-side view when assuming the server is in tcp_ecn=2 mode, that is, Linux default since 2009 via commit 255cac91c3c9 ("tcp: extend ECN sysctl to allow server-side only ECN"): 1) Normal ECN-capable path: SYN ECE CWR -----> <----- SYN ACK ECE ACK -----> 2) Path with broken middlebox, when client has fallback: SYN ECE CWR ----X crappy middlebox drops packet (timeout, rtx) SYN -----> <----- SYN ACK ACK -----> In case we would not have the fallback implemented, the middlebox drop point would basically end up as: SYN ECE CWR ----X crappy middlebox drops packet (timeout, rtx) SYN ECE CWR ----X crappy middlebox drops packet (timeout, rtx) SYN ECE CWR ----X crappy middlebox drops packet (timeout, rtx) In any case, it's rather a smaller percentage of sites where there would occur such additional setup latency: it was found in end of 2014 that ~56% of IPv4 and 65% of IPv6 servers of Alexa 1 million list would negotiate ECN (aka tcp_ecn=2 default), 0.42% of these webservers will fail to connect when trying to negotiate with ECN (tcp_ecn=1) due to timeouts, which the fallback would mitigate with a slight latency trade-off. Recent related paper on this topic: Brian Trammell, Mirja Kühlewind, Damiano Boppart, Iain Learmonth, Gorry Fairhurst, and Richard Scheffenegger: "Enabling Internet-Wide Deployment of Explicit Congestion Notification." Proc. PAM 2015, New York. http://ecn.ethz.ch/ecn-pam15.pdf Thus, when net.ipv4.tcp_ecn=1 is being set, the patch will perform RFC3168, section 6.1.1.1. fallback on timeout. For users explicitly not wanting this which can be in DC use case, we add a net.ipv4.tcp_ecn_fallback knob that allows for disabling the fallback. tp->ecn_flags are not being cleared in tcp_ecn_clear_syn() on output, but rather we let tcp_ecn_rcv_synack() take that over on input path in case a SYN ACK ECE was delayed. Thus a spurious SYN retransmission will not prevent ECN being negotiated eventually in that case. Reference: https://www.ietf.org/proceedings/92/slides/slides-92-iccrg-1.pdf Reference: https://www.ietf.org/proceedings/89/slides/slides-89-tsvarea-1.pdf Signed-off-by: Daniel Borkmann <daniel@iogearbox.net> Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: Mirja Kühlewind <mirja.kuehlewind@tik.ee.ethz.ch> Signed-off-by: Brian Trammell <trammell@tik.ee.ethz.ch> Cc: Eric Dumazet <edumazet@google.com> Cc: Dave That <dave.taht@gmail.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-05-20 03:04:22 +08:00
#define TCPHDR_SYN_ECN (TCPHDR_SYN | TCPHDR_ECE | TCPHDR_CWR)
/* This is what the send packet queuing engine uses to pass
* TCP per-packet control information to the transmission code.
* We also store the host-order sequence numbers in here too.
* This is 44 bytes if IPV6 is enabled.
* If this grows please adjust skbuff.h:skbuff->cb[xxx] size appropriately.
*/
struct tcp_skb_cb {
__u32 seq; /* Starting sequence number */
__u32 end_seq; /* SEQ + FIN + SYN + datalen */
union {
/* Note : tcp_tw_isn is used in input path only
* (isn chosen by tcp_timewait_state_process())
*
* tcp_gso_segs/size are used in write queue only,
* cf tcp_skb_pcount()/tcp_skb_mss()
*/
__u32 tcp_tw_isn;
struct {
u16 tcp_gso_segs;
u16 tcp_gso_size;
};
};
__u8 tcp_flags; /* TCP header flags. (tcp[13]) */
__u8 sacked; /* State flags for SACK. */
#define TCPCB_SACKED_ACKED 0x01 /* SKB ACK'd by a SACK block */
#define TCPCB_SACKED_RETRANS 0x02 /* SKB retransmitted */
#define TCPCB_LOST 0x04 /* SKB is lost */
#define TCPCB_TAGBITS 0x07 /* All tag bits */
#define TCPCB_REPAIRED 0x10 /* SKB repaired (no skb_mstamp_ns) */
#define TCPCB_EVER_RETRANS 0x80 /* Ever retransmitted frame */
tcp: don't use timestamp from repaired skb-s to calculate RTT (v2) We don't know right timestamp for repaired skb-s. Wrong RTT estimations isn't good, because some congestion modules heavily depends on it. This patch adds the TCPCB_REPAIRED flag, which is included in TCPCB_RETRANS. Thanks to Eric for the advice how to fix this issue. This patch fixes the warning: [ 879.562947] WARNING: CPU: 0 PID: 2825 at net/ipv4/tcp_input.c:3078 tcp_ack+0x11f5/0x1380() [ 879.567253] CPU: 0 PID: 2825 Comm: socket-tcpbuf-l Not tainted 3.16.0-next-20140811 #1 [ 879.567829] Hardware name: Bochs Bochs, BIOS Bochs 01/01/2011 [ 879.568177] 0000000000000000 00000000c532680c ffff880039643d00 ffffffff817aa2d2 [ 879.568776] 0000000000000000 ffff880039643d38 ffffffff8109afbd ffff880039d6ba80 [ 879.569386] ffff88003a449800 000000002983d6bd 0000000000000000 000000002983d6bc [ 879.569982] Call Trace: [ 879.570264] [<ffffffff817aa2d2>] dump_stack+0x4d/0x66 [ 879.570599] [<ffffffff8109afbd>] warn_slowpath_common+0x7d/0xa0 [ 879.570935] [<ffffffff8109b0ea>] warn_slowpath_null+0x1a/0x20 [ 879.571292] [<ffffffff816d0a05>] tcp_ack+0x11f5/0x1380 [ 879.571614] [<ffffffff816d10bd>] tcp_rcv_established+0x1ed/0x710 [ 879.571958] [<ffffffff816dc9da>] tcp_v4_do_rcv+0x10a/0x370 [ 879.572315] [<ffffffff81657459>] release_sock+0x89/0x1d0 [ 879.572642] [<ffffffff816c81a0>] do_tcp_setsockopt.isra.36+0x120/0x860 [ 879.573000] [<ffffffff8110a52e>] ? rcu_read_lock_held+0x6e/0x80 [ 879.573352] [<ffffffff816c8912>] tcp_setsockopt+0x32/0x40 [ 879.573678] [<ffffffff81654ac4>] sock_common_setsockopt+0x14/0x20 [ 879.574031] [<ffffffff816537b0>] SyS_setsockopt+0x80/0xf0 [ 879.574393] [<ffffffff817b40a9>] system_call_fastpath+0x16/0x1b [ 879.574730] ---[ end trace a17cbc38eb8c5c00 ]--- v2: moving setting of skb->when for repaired skb-s in tcp_write_xmit, where it's set for other skb-s. Fixes: 431a91242d8d ("tcp: timestamp SYN+DATA messages") Fixes: 740b0f1841f6 ("tcp: switch rtt estimations to usec resolution") Cc: Eric Dumazet <edumazet@google.com> Cc: Pavel Emelyanov <xemul@parallels.com> Cc: "David S. Miller" <davem@davemloft.net> Signed-off-by: Andrey Vagin <avagin@openvz.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-08-13 20:03:10 +08:00
#define TCPCB_RETRANS (TCPCB_SACKED_RETRANS|TCPCB_EVER_RETRANS| \
TCPCB_REPAIRED)
__u8 ip_dsfield; /* IPv4 tos or IPv6 dsfield */
__u8 txstamp_ack:1, /* Record TX timestamp for ack? */
tcp: Make use of MSG_EOR in tcp_sendmsg This patch adds an eor bit to the TCP_SKB_CB. When MSG_EOR is passed to tcp_sendmsg, the eor bit will be set at the skb containing the last byte of the userland's msg. The eor bit will prevent data from appending to that skb in the future. The change in do_tcp_sendpages is to honor the eor set during the previous tcp_sendmsg(MSG_EOR) call. This patch handles the tcp_sendmsg case. The followup patches will handle other skb coalescing and fragment cases. One potential use case is to use MSG_EOR with SOF_TIMESTAMPING_TX_ACK to get a more accurate TCP ack timestamping on application protocol with multiple outgoing response messages (e.g. HTTP2). Packetdrill script for testing: ~~~~~~ +0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10` +0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1` +0 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7> 0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7> 0.200 < . 1:1(0) ack 1 win 257 0.200 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0 0.200 write(4, ..., 14600) = 14600 0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730 0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730 0.200 > . 1:7301(7300) ack 1 0.200 > P. 7301:14601(7300) ack 1 0.300 < . 1:1(0) ack 14601 win 257 0.300 > P. 14601:15331(730) ack 1 0.300 > P. 15331:16061(730) ack 1 0.400 < . 1:1(0) ack 16061 win 257 0.400 close(4) = 0 0.400 > F. 16061:16061(0) ack 1 0.400 < F. 1:1(0) ack 16062 win 257 0.400 > . 16062:16062(0) ack 2 Signed-off-by: Martin KaFai Lau <kafai@fb.com> Cc: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Soheil Hassas Yeganeh <soheil@google.com> Cc: Willem de Bruijn <willemb@google.com> Cc: Yuchung Cheng <ycheng@google.com> Suggested-by: Eric Dumazet <edumazet@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-04-26 05:44:48 +08:00
eor:1, /* Is skb MSG_EOR marked? */
has_rxtstamp:1, /* SKB has a RX timestamp */
unused:5;
__u32 ack_seq; /* Sequence number ACK'd */
union {
struct {
tcp: track data delivery rate for a TCP connection This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
/* There is space for up to 24 bytes */
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 11:39:15 +08:00
__u32 in_flight:30,/* Bytes in flight at transmit */
is_app_limited:1, /* cwnd not fully used? */
unused:1;
tcp: track data delivery rate for a TCP connection This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
/* pkts S/ACKed so far upon tx of skb, incl retrans: */
__u32 delivered;
/* start of send pipeline phase */
u64 first_tx_mstamp;
tcp: track data delivery rate for a TCP connection This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
/* when we reached the "delivered" count */
u64 delivered_mstamp;
} tx; /* only used for outgoing skbs */
union {
struct inet_skb_parm h4;
#if IS_ENABLED(CONFIG_IPV6)
struct inet6_skb_parm h6;
#endif
} header; /* For incoming skbs */
struct {
__u32 flags;
struct sock *sk_redir;
void *data_end;
} bpf;
};
};
#define TCP_SKB_CB(__skb) ((struct tcp_skb_cb *)&((__skb)->cb[0]))
bpf: sockmap, convert bpf_compute_data_pointers to bpf_*_sk_skb In commit 'bpf: bpf_compute_data uses incorrect cb structure' (8108a7751512) we added the routine bpf_compute_data_end_sk_skb() to compute the correct data_end values, but this has since been lost. In kernel v4.14 this was correct and the above patch was applied in it entirety. Then when v4.14 was merged into v4.15-rc1 net-next tree we lost the piece that renamed bpf_compute_data_pointers to the new function bpf_compute_data_end_sk_skb. This was done here, e1ea2f9856b7 ("Merge git://git.kernel.org/pub/scm/linux/kernel/git/davem/net") When it conflicted with the following rename patch, 6aaae2b6c433 ("bpf: rename bpf_compute_data_end into bpf_compute_data_pointers") Finally, after a refactor I thought even the function bpf_compute_data_end_sk_skb() was no longer needed and it was erroneously removed. However, we never reverted the sk_skb_convert_ctx_access() usage of tcp_skb_cb which had been committed and survived the merge conflict. Here we fix this by adding back the helper and *_data_end_sk_skb() usage. Using the bpf_skc_data_end mapping is not correct because it expects a qdisc_skb_cb object but at the sock layer this is not the case. Even though it happens to work here because we don't overwrite any data in-use at the socket layer and the cb structure is cleared later this has potential to create some subtle issues. But, even more concretely the filter.c access check uses tcp_skb_cb. And by some act of chance though, struct bpf_skb_data_end { struct qdisc_skb_cb qdisc_cb; /* 0 28 */ /* XXX 4 bytes hole, try to pack */ void * data_meta; /* 32 8 */ void * data_end; /* 40 8 */ /* size: 48, cachelines: 1, members: 3 */ /* sum members: 44, holes: 1, sum holes: 4 */ /* last cacheline: 48 bytes */ }; and then tcp_skb_cb, struct tcp_skb_cb { [...] struct { __u32 flags; /* 24 4 */ struct sock * sk_redir; /* 32 8 */ void * data_end; /* 40 8 */ } bpf; /* 24 */ }; So when we use offset_of() to track down the byte offset we get 40 in either case and everything continues to work. Fix this mess and use correct structures its unclear how long this might actually work for until someone moves the structs around. Reported-by: Martin KaFai Lau <kafai@fb.com> Fixes: e1ea2f9856b7 ("Merge git://git.kernel.org/pub/scm/linux/kernel/git/davem/net") Fixes: 6aaae2b6c433 ("bpf: rename bpf_compute_data_end into bpf_compute_data_pointers") Signed-off-by: John Fastabend <john.fastabend@gmail.com> Signed-off-by: Alexei Starovoitov <ast@kernel.org>
2018-07-05 23:50:15 +08:00
static inline void bpf_compute_data_end_sk_skb(struct sk_buff *skb)
{
TCP_SKB_CB(skb)->bpf.data_end = skb->data + skb_headlen(skb);
}
bpf, sockmap: convert to generic sk_msg interface Add a generic sk_msg layer, and convert current sockmap and later kTLS over to make use of it. While sk_buff handles network packet representation from netdevice up to socket, sk_msg handles data representation from application to socket layer. This means that sk_msg framework spans across ULP users in the kernel, and enables features such as introspection or filtering of data with the help of BPF programs that operate on this data structure. Latter becomes in particular useful for kTLS where data encryption is deferred into the kernel, and as such enabling the kernel to perform L7 introspection and policy based on BPF for TLS connections where the record is being encrypted after BPF has run and came to a verdict. In order to get there, first step is to transform open coding of scatter-gather list handling into a common core framework that subsystems can use. The code itself has been split and refactored into three bigger pieces: i) the generic sk_msg API which deals with managing the scatter gather ring, providing helpers for walking and mangling, transferring application data from user space into it, and preparing it for BPF pre/post-processing, ii) the plain sock map itself where sockets can be attached to or detached from; these bits are independent of i) which can now be used also without sock map, and iii) the integration with plain TCP as one protocol to be used for processing L7 application data (later this could e.g. also be extended to other protocols like UDP). The semantics are the same with the old sock map code and therefore no change of user facing behavior or APIs. While pursuing this work it also helped finding a number of bugs in the old sockmap code that we've fixed already in earlier commits. The test_sockmap kselftest suite passes through fine as well. Joint work with John. Signed-off-by: Daniel Borkmann <daniel@iogearbox.net> Signed-off-by: John Fastabend <john.fastabend@gmail.com> Signed-off-by: Alexei Starovoitov <ast@kernel.org>
2018-10-13 08:45:58 +08:00
static inline bool tcp_skb_bpf_ingress(const struct sk_buff *skb)
{
return TCP_SKB_CB(skb)->bpf.flags & BPF_F_INGRESS;
}
static inline struct sock *tcp_skb_bpf_redirect_fetch(struct sk_buff *skb)
{
return TCP_SKB_CB(skb)->bpf.sk_redir;
}
static inline void tcp_skb_bpf_redirect_clear(struct sk_buff *skb)
{
TCP_SKB_CB(skb)->bpf.sk_redir = NULL;
}
extern const struct inet_connection_sock_af_ops ipv4_specific;
#if IS_ENABLED(CONFIG_IPV6)
/* This is the variant of inet6_iif() that must be used by TCP,
* as TCP moves IP6CB into a different location in skb->cb[]
*/
static inline int tcp_v6_iif(const struct sk_buff *skb)
{
return TCP_SKB_CB(skb)->header.h6.iif;
}
static inline int tcp_v6_iif_l3_slave(const struct sk_buff *skb)
{
net: Require exact match for TCP socket lookups if dif is l3mdev Currently, socket lookups for l3mdev (vrf) use cases can match a socket that is bound to a port but not a device (ie., a global socket). If the sysctl tcp_l3mdev_accept is not set this leads to ack packets going out based on the main table even though the packet came in from an L3 domain. The end result is that the connection does not establish creating confusion for users since the service is running and a socket shows in ss output. Fix by requiring an exact dif to sk_bound_dev_if match if the skb came through an interface enslaved to an l3mdev device and the tcp_l3mdev_accept is not set. skb's through an l3mdev interface are marked by setting a flag in inet{6}_skb_parm. The IPv6 variant is already set; this patch adds the flag for IPv4. Using an skb flag avoids a device lookup on the dif. The flag is set in the VRF driver using the IP{6}CB macros. For IPv4, the inet_skb_parm struct is moved in the cb per commit 971f10eca186, so the match function in the TCP stack needs to use TCP_SKB_CB. For IPv6, the move is done after the socket lookup, so IP6CB is used. The flags field in inet_skb_parm struct needs to be increased to add another flag. There is currently a 1-byte hole following the flags, so it can be expanded to u16 without increasing the size of the struct. Fixes: 193125dbd8eb ("net: Introduce VRF device driver") Signed-off-by: David Ahern <dsa@cumulusnetworks.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-10-17 11:02:52 +08:00
bool l3_slave = ipv6_l3mdev_skb(TCP_SKB_CB(skb)->header.h6.flags);
return l3_slave ? skb->skb_iif : TCP_SKB_CB(skb)->header.h6.iif;
}
/* TCP_SKB_CB reference means this can not be used from early demux */
static inline int tcp_v6_sdif(const struct sk_buff *skb)
{
#if IS_ENABLED(CONFIG_NET_L3_MASTER_DEV)
if (skb && ipv6_l3mdev_skb(TCP_SKB_CB(skb)->header.h6.flags))
return TCP_SKB_CB(skb)->header.h6.iif;
#endif
return 0;
}
extern const struct inet_connection_sock_af_ops ipv6_specific;
INDIRECT_CALLABLE_DECLARE(void tcp_v6_send_check(struct sock *sk, struct sk_buff *skb));
INDIRECT_CALLABLE_DECLARE(int tcp_v6_rcv(struct sk_buff *skb));
INDIRECT_CALLABLE_DECLARE(void tcp_v6_early_demux(struct sk_buff *skb));
#endif
/* TCP_SKB_CB reference means this can not be used from early demux */
static inline int tcp_v4_sdif(struct sk_buff *skb)
{
#if IS_ENABLED(CONFIG_NET_L3_MASTER_DEV)
if (skb && ipv4_l3mdev_skb(TCP_SKB_CB(skb)->header.h4.flags))
return TCP_SKB_CB(skb)->header.h4.iif;
#endif
return 0;
}
/* Due to TSO, an SKB can be composed of multiple actual
* packets. To keep these tracked properly, we use this.
*/
static inline int tcp_skb_pcount(const struct sk_buff *skb)
{
return TCP_SKB_CB(skb)->tcp_gso_segs;
}
static inline void tcp_skb_pcount_set(struct sk_buff *skb, int segs)
{
TCP_SKB_CB(skb)->tcp_gso_segs = segs;
}
static inline void tcp_skb_pcount_add(struct sk_buff *skb, int segs)
{
TCP_SKB_CB(skb)->tcp_gso_segs += segs;
}
/* This is valid iff skb is in write queue and tcp_skb_pcount() > 1. */
static inline int tcp_skb_mss(const struct sk_buff *skb)
{
return TCP_SKB_CB(skb)->tcp_gso_size;
}
tcp: Make use of MSG_EOR in tcp_sendmsg This patch adds an eor bit to the TCP_SKB_CB. When MSG_EOR is passed to tcp_sendmsg, the eor bit will be set at the skb containing the last byte of the userland's msg. The eor bit will prevent data from appending to that skb in the future. The change in do_tcp_sendpages is to honor the eor set during the previous tcp_sendmsg(MSG_EOR) call. This patch handles the tcp_sendmsg case. The followup patches will handle other skb coalescing and fragment cases. One potential use case is to use MSG_EOR with SOF_TIMESTAMPING_TX_ACK to get a more accurate TCP ack timestamping on application protocol with multiple outgoing response messages (e.g. HTTP2). Packetdrill script for testing: ~~~~~~ +0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10` +0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1` +0 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7> 0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7> 0.200 < . 1:1(0) ack 1 win 257 0.200 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0 0.200 write(4, ..., 14600) = 14600 0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730 0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730 0.200 > . 1:7301(7300) ack 1 0.200 > P. 7301:14601(7300) ack 1 0.300 < . 1:1(0) ack 14601 win 257 0.300 > P. 14601:15331(730) ack 1 0.300 > P. 15331:16061(730) ack 1 0.400 < . 1:1(0) ack 16061 win 257 0.400 close(4) = 0 0.400 > F. 16061:16061(0) ack 1 0.400 < F. 1:1(0) ack 16062 win 257 0.400 > . 16062:16062(0) ack 2 Signed-off-by: Martin KaFai Lau <kafai@fb.com> Cc: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Soheil Hassas Yeganeh <soheil@google.com> Cc: Willem de Bruijn <willemb@google.com> Cc: Yuchung Cheng <ycheng@google.com> Suggested-by: Eric Dumazet <edumazet@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-04-26 05:44:48 +08:00
static inline bool tcp_skb_can_collapse_to(const struct sk_buff *skb)
{
return likely(!TCP_SKB_CB(skb)->eor);
}
static inline bool tcp_skb_can_collapse(const struct sk_buff *to,
const struct sk_buff *from)
{
return likely(tcp_skb_can_collapse_to(to) &&
mptcp_skb_can_collapse(to, from));
}
/* Events passed to congestion control interface */
enum tcp_ca_event {
CA_EVENT_TX_START, /* first transmit when no packets in flight */
CA_EVENT_CWND_RESTART, /* congestion window restart */
CA_EVENT_COMPLETE_CWR, /* end of congestion recovery */
CA_EVENT_LOSS, /* loss timeout */
CA_EVENT_ECN_NO_CE, /* ECT set, but not CE marked */
CA_EVENT_ECN_IS_CE, /* received CE marked IP packet */
};
/* Information about inbound ACK, passed to cong_ops->in_ack_event() */
enum tcp_ca_ack_event_flags {
CA_ACK_SLOWPATH = (1 << 0), /* In slow path processing */
CA_ACK_WIN_UPDATE = (1 << 1), /* ACK updated window */
CA_ACK_ECE = (1 << 2), /* ECE bit is set on ack */
};
/*
* Interface for adding new TCP congestion control handlers
*/
#define TCP_CA_NAME_MAX 16
#define TCP_CA_MAX 128
#define TCP_CA_BUF_MAX (TCP_CA_NAME_MAX*TCP_CA_MAX)
net: tcp: add key management to congestion control This patch adds necessary infrastructure to the congestion control framework for later per route congestion control support. For a per route congestion control possibility, our aim is to store a unique u32 key identifier into dst metrics, which can then be mapped into a tcp_congestion_ops struct. We argue that having a RTAX key entry is the most simple, generic and easy way to manage, and also keeps the memory footprint of dst entries lower on 64 bit than with storing a pointer directly, for example. Having a unique key id also allows for decoupling actual TCP congestion control module management from the FIB layer, i.e. we don't have to care about expensive module refcounting inside the FIB at this point. We first thought of using an IDR store for the realization, which takes over dynamic assignment of unused key space and also performs the key to pointer mapping in RCU. While doing so, we stumbled upon the issue that due to the nature of dynamic key distribution, it just so happens, arguably in very rare occasions, that excessive module loads and unloads can lead to a possible reuse of previously used key space. Thus, previously stale keys in the dst metric are now being reassigned to a different congestion control algorithm, which might lead to unexpected behaviour. One way to resolve this would have been to walk FIBs on the actually rare occasion of a module unload and reset the metric keys for each FIB in each netns, but that's just very costly. Therefore, we argue a better solution is to reuse the unique congestion control algorithm name member and map that into u32 key space through jhash. For that, we split the flags attribute (as it currently uses 2 bits only anyway) into two u32 attributes, flags and key, so that we can keep the cacheline boundary of 2 cachelines on x86_64 and cache the precalculated key at registration time for the fast path. On average we might expect 2 - 4 modules being loaded worst case perhaps 15, so a key collision possibility is extremely low, and guaranteed collision-free on LE/BE for all in-tree modules. Overall this results in much simpler code, and all without the overhead of an IDR. Due to the deterministic nature, modules can now be unloaded, the congestion control algorithm for a specific but unloaded key will fall back to the default one, and on module reload time it will switch back to the expected algorithm transparently. Joint work with Florian Westphal. Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: Daniel Borkmann <dborkman@redhat.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:46 +08:00
#define TCP_CA_UNSPEC 0
/* Algorithm can be set on socket without CAP_NET_ADMIN privileges */
#define TCP_CONG_NON_RESTRICTED 0x1
/* Requires ECN/ECT set on all packets */
#define TCP_CONG_NEEDS_ECN 0x2
bpf: tcp: Support tcp_congestion_ops in bpf This patch makes "struct tcp_congestion_ops" to be the first user of BPF STRUCT_OPS. It allows implementing a tcp_congestion_ops in bpf. The BPF implemented tcp_congestion_ops can be used like regular kernel tcp-cc through sysctl and setsockopt. e.g. [root@arch-fb-vm1 bpf]# sysctl -a | egrep congestion net.ipv4.tcp_allowed_congestion_control = reno cubic bpf_cubic net.ipv4.tcp_available_congestion_control = reno bic cubic bpf_cubic net.ipv4.tcp_congestion_control = bpf_cubic There has been attempt to move the TCP CC to the user space (e.g. CCP in TCP). The common arguments are faster turn around, get away from long-tail kernel versions in production...etc, which are legit points. BPF has been the continuous effort to join both kernel and userspace upsides together (e.g. XDP to gain the performance advantage without bypassing the kernel). The recent BPF advancements (in particular BTF-aware verifier, BPF trampoline, BPF CO-RE...) made implementing kernel struct ops (e.g. tcp cc) possible in BPF. It allows a faster turnaround for testing algorithm in the production while leveraging the existing (and continue growing) BPF feature/framework instead of building one specifically for userspace TCP CC. This patch allows write access to a few fields in tcp-sock (in bpf_tcp_ca_btf_struct_access()). The optional "get_info" is unsupported now. It can be added later. One possible way is to output the info with a btf-id to describe the content. Signed-off-by: Martin KaFai Lau <kafai@fb.com> Signed-off-by: Alexei Starovoitov <ast@kernel.org> Acked-by: Andrii Nakryiko <andriin@fb.com> Acked-by: Yonghong Song <yhs@fb.com> Link: https://lore.kernel.org/bpf/20200109003508.3856115-1-kafai@fb.com
2020-01-09 08:35:08 +08:00
#define TCP_CONG_MASK (TCP_CONG_NON_RESTRICTED | TCP_CONG_NEEDS_ECN)
union tcp_cc_info;
struct ack_sample {
u32 pkts_acked;
s32 rtt_us;
u32 in_flight;
};
tcp: track data delivery rate for a TCP connection This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
/* A rate sample measures the number of (original/retransmitted) data
* packets delivered "delivered" over an interval of time "interval_us".
* The tcp_rate.c code fills in the rate sample, and congestion
* control modules that define a cong_control function to run at the end
* of ACK processing can optionally chose to consult this sample when
* setting cwnd and pacing rate.
* A sample is invalid if "delivered" or "interval_us" is negative.
*/
struct rate_sample {
u64 prior_mstamp; /* starting timestamp for interval */
tcp: track data delivery rate for a TCP connection This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
u32 prior_delivered; /* tp->delivered at "prior_mstamp" */
s32 delivered; /* number of packets delivered over interval */
long interval_us; /* time for tp->delivered to incr "delivered" */
u32 snd_interval_us; /* snd interval for delivered packets */
u32 rcv_interval_us; /* rcv interval for delivered packets */
tcp: track data delivery rate for a TCP connection This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
long rtt_us; /* RTT of last (S)ACKed packet (or -1) */
int losses; /* number of packets marked lost upon ACK */
u32 acked_sacked; /* number of packets newly (S)ACKed upon ACK */
u32 prior_in_flight; /* in flight before this ACK */
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 11:39:15 +08:00
bool is_app_limited; /* is sample from packet with bubble in pipe? */
tcp: track data delivery rate for a TCP connection This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
bool is_retrans; /* is sample from retransmission? */
bool is_ack_delayed; /* is this (likely) a delayed ACK? */
tcp: track data delivery rate for a TCP connection This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
};
struct tcp_congestion_ops {
struct list_head list;
net: tcp: add key management to congestion control This patch adds necessary infrastructure to the congestion control framework for later per route congestion control support. For a per route congestion control possibility, our aim is to store a unique u32 key identifier into dst metrics, which can then be mapped into a tcp_congestion_ops struct. We argue that having a RTAX key entry is the most simple, generic and easy way to manage, and also keeps the memory footprint of dst entries lower on 64 bit than with storing a pointer directly, for example. Having a unique key id also allows for decoupling actual TCP congestion control module management from the FIB layer, i.e. we don't have to care about expensive module refcounting inside the FIB at this point. We first thought of using an IDR store for the realization, which takes over dynamic assignment of unused key space and also performs the key to pointer mapping in RCU. While doing so, we stumbled upon the issue that due to the nature of dynamic key distribution, it just so happens, arguably in very rare occasions, that excessive module loads and unloads can lead to a possible reuse of previously used key space. Thus, previously stale keys in the dst metric are now being reassigned to a different congestion control algorithm, which might lead to unexpected behaviour. One way to resolve this would have been to walk FIBs on the actually rare occasion of a module unload and reset the metric keys for each FIB in each netns, but that's just very costly. Therefore, we argue a better solution is to reuse the unique congestion control algorithm name member and map that into u32 key space through jhash. For that, we split the flags attribute (as it currently uses 2 bits only anyway) into two u32 attributes, flags and key, so that we can keep the cacheline boundary of 2 cachelines on x86_64 and cache the precalculated key at registration time for the fast path. On average we might expect 2 - 4 modules being loaded worst case perhaps 15, so a key collision possibility is extremely low, and guaranteed collision-free on LE/BE for all in-tree modules. Overall this results in much simpler code, and all without the overhead of an IDR. Due to the deterministic nature, modules can now be unloaded, the congestion control algorithm for a specific but unloaded key will fall back to the default one, and on module reload time it will switch back to the expected algorithm transparently. Joint work with Florian Westphal. Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: Daniel Borkmann <dborkman@redhat.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:46 +08:00
u32 key;
u32 flags;
/* initialize private data (optional) */
void (*init)(struct sock *sk);
/* cleanup private data (optional) */
void (*release)(struct sock *sk);
/* return slow start threshold (required) */
u32 (*ssthresh)(struct sock *sk);
/* do new cwnd calculation (required) */
void (*cong_avoid)(struct sock *sk, u32 ack, u32 acked);
/* call before changing ca_state (optional) */
void (*set_state)(struct sock *sk, u8 new_state);
/* call when cwnd event occurs (optional) */
void (*cwnd_event)(struct sock *sk, enum tcp_ca_event ev);
/* call when ack arrives (optional) */
void (*in_ack_event)(struct sock *sk, u32 flags);
/* new value of cwnd after loss (required) */
u32 (*undo_cwnd)(struct sock *sk);
/* hook for packet ack accounting (optional) */
void (*pkts_acked)(struct sock *sk, const struct ack_sample *sample);
/* override sysctl_tcp_min_tso_segs */
u32 (*min_tso_segs)(struct sock *sk);
/* returns the multiplier used in tcp_sndbuf_expand (optional) */
u32 (*sndbuf_expand)(struct sock *sk);
tcp: new CC hook to set sending rate with rate_sample in any CA state This commit introduces an optional new "omnipotent" hook, cong_control(), for congestion control modules. The cong_control() function is called at the end of processing an ACK (i.e., after updating sequence numbers, the SACK scoreboard, and loss detection). At that moment we have precise delivery rate information the congestion control module can use to control the sending behavior (using cwnd, TSO skb size, and pacing rate) in any CA state. This function can also be used by a congestion control that prefers not to use the default cwnd reduction approach (i.e., the PRR algorithm) during CA_Recovery to control the cwnd and sending rate during loss recovery. We take advantage of the fact that recent changes defer the retransmission or transmission of new data (e.g. by F-RTO) in recovery until the new tcp_cong_control() function is run. With this commit, we only run tcp_update_pacing_rate() if the congestion control is not using this new API. New congestion controls which use the new API do not want the TCP stack to run the default pacing rate calculation and overwrite whatever pacing rate they have chosen at initialization time. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:21 +08:00
/* call when packets are delivered to update cwnd and pacing rate,
* after all the ca_state processing. (optional)
*/
void (*cong_control)(struct sock *sk, const struct rate_sample *rs);
/* get info for inet_diag (optional) */
size_t (*get_info)(struct sock *sk, u32 ext, int *attr,
union tcp_cc_info *info);
char name[TCP_CA_NAME_MAX];
struct module *owner;
};
int tcp_register_congestion_control(struct tcp_congestion_ops *type);
void tcp_unregister_congestion_control(struct tcp_congestion_ops *type);
void tcp_assign_congestion_control(struct sock *sk);
void tcp_init_congestion_control(struct sock *sk);
void tcp_cleanup_congestion_control(struct sock *sk);
int tcp_set_default_congestion_control(struct net *net, const char *name);
void tcp_get_default_congestion_control(struct net *net, char *name);
void tcp_get_available_congestion_control(char *buf, size_t len);
void tcp_get_allowed_congestion_control(char *buf, size_t len);
int tcp_set_allowed_congestion_control(char *allowed);
int tcp_set_congestion_control(struct sock *sk, const char *name, bool load,
bool cap_net_admin);
u32 tcp_slow_start(struct tcp_sock *tp, u32 acked);
void tcp_cong_avoid_ai(struct tcp_sock *tp, u32 w, u32 acked);
u32 tcp_reno_ssthresh(struct sock *sk);
u32 tcp_reno_undo_cwnd(struct sock *sk);
void tcp_reno_cong_avoid(struct sock *sk, u32 ack, u32 acked);
extern struct tcp_congestion_ops tcp_reno;
bpf: tcp: Support tcp_congestion_ops in bpf This patch makes "struct tcp_congestion_ops" to be the first user of BPF STRUCT_OPS. It allows implementing a tcp_congestion_ops in bpf. The BPF implemented tcp_congestion_ops can be used like regular kernel tcp-cc through sysctl and setsockopt. e.g. [root@arch-fb-vm1 bpf]# sysctl -a | egrep congestion net.ipv4.tcp_allowed_congestion_control = reno cubic bpf_cubic net.ipv4.tcp_available_congestion_control = reno bic cubic bpf_cubic net.ipv4.tcp_congestion_control = bpf_cubic There has been attempt to move the TCP CC to the user space (e.g. CCP in TCP). The common arguments are faster turn around, get away from long-tail kernel versions in production...etc, which are legit points. BPF has been the continuous effort to join both kernel and userspace upsides together (e.g. XDP to gain the performance advantage without bypassing the kernel). The recent BPF advancements (in particular BTF-aware verifier, BPF trampoline, BPF CO-RE...) made implementing kernel struct ops (e.g. tcp cc) possible in BPF. It allows a faster turnaround for testing algorithm in the production while leveraging the existing (and continue growing) BPF feature/framework instead of building one specifically for userspace TCP CC. This patch allows write access to a few fields in tcp-sock (in bpf_tcp_ca_btf_struct_access()). The optional "get_info" is unsupported now. It can be added later. One possible way is to output the info with a btf-id to describe the content. Signed-off-by: Martin KaFai Lau <kafai@fb.com> Signed-off-by: Alexei Starovoitov <ast@kernel.org> Acked-by: Andrii Nakryiko <andriin@fb.com> Acked-by: Yonghong Song <yhs@fb.com> Link: https://lore.kernel.org/bpf/20200109003508.3856115-1-kafai@fb.com
2020-01-09 08:35:08 +08:00
struct tcp_congestion_ops *tcp_ca_find(const char *name);
net: tcp: add key management to congestion control This patch adds necessary infrastructure to the congestion control framework for later per route congestion control support. For a per route congestion control possibility, our aim is to store a unique u32 key identifier into dst metrics, which can then be mapped into a tcp_congestion_ops struct. We argue that having a RTAX key entry is the most simple, generic and easy way to manage, and also keeps the memory footprint of dst entries lower on 64 bit than with storing a pointer directly, for example. Having a unique key id also allows for decoupling actual TCP congestion control module management from the FIB layer, i.e. we don't have to care about expensive module refcounting inside the FIB at this point. We first thought of using an IDR store for the realization, which takes over dynamic assignment of unused key space and also performs the key to pointer mapping in RCU. While doing so, we stumbled upon the issue that due to the nature of dynamic key distribution, it just so happens, arguably in very rare occasions, that excessive module loads and unloads can lead to a possible reuse of previously used key space. Thus, previously stale keys in the dst metric are now being reassigned to a different congestion control algorithm, which might lead to unexpected behaviour. One way to resolve this would have been to walk FIBs on the actually rare occasion of a module unload and reset the metric keys for each FIB in each netns, but that's just very costly. Therefore, we argue a better solution is to reuse the unique congestion control algorithm name member and map that into u32 key space through jhash. For that, we split the flags attribute (as it currently uses 2 bits only anyway) into two u32 attributes, flags and key, so that we can keep the cacheline boundary of 2 cachelines on x86_64 and cache the precalculated key at registration time for the fast path. On average we might expect 2 - 4 modules being loaded worst case perhaps 15, so a key collision possibility is extremely low, and guaranteed collision-free on LE/BE for all in-tree modules. Overall this results in much simpler code, and all without the overhead of an IDR. Due to the deterministic nature, modules can now be unloaded, the congestion control algorithm for a specific but unloaded key will fall back to the default one, and on module reload time it will switch back to the expected algorithm transparently. Joint work with Florian Westphal. Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: Daniel Borkmann <dborkman@redhat.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:46 +08:00
struct tcp_congestion_ops *tcp_ca_find_key(u32 key);
u32 tcp_ca_get_key_by_name(struct net *net, const char *name, bool *ecn_ca);
#ifdef CONFIG_INET
net: tcp: add key management to congestion control This patch adds necessary infrastructure to the congestion control framework for later per route congestion control support. For a per route congestion control possibility, our aim is to store a unique u32 key identifier into dst metrics, which can then be mapped into a tcp_congestion_ops struct. We argue that having a RTAX key entry is the most simple, generic and easy way to manage, and also keeps the memory footprint of dst entries lower on 64 bit than with storing a pointer directly, for example. Having a unique key id also allows for decoupling actual TCP congestion control module management from the FIB layer, i.e. we don't have to care about expensive module refcounting inside the FIB at this point. We first thought of using an IDR store for the realization, which takes over dynamic assignment of unused key space and also performs the key to pointer mapping in RCU. While doing so, we stumbled upon the issue that due to the nature of dynamic key distribution, it just so happens, arguably in very rare occasions, that excessive module loads and unloads can lead to a possible reuse of previously used key space. Thus, previously stale keys in the dst metric are now being reassigned to a different congestion control algorithm, which might lead to unexpected behaviour. One way to resolve this would have been to walk FIBs on the actually rare occasion of a module unload and reset the metric keys for each FIB in each netns, but that's just very costly. Therefore, we argue a better solution is to reuse the unique congestion control algorithm name member and map that into u32 key space through jhash. For that, we split the flags attribute (as it currently uses 2 bits only anyway) into two u32 attributes, flags and key, so that we can keep the cacheline boundary of 2 cachelines on x86_64 and cache the precalculated key at registration time for the fast path. On average we might expect 2 - 4 modules being loaded worst case perhaps 15, so a key collision possibility is extremely low, and guaranteed collision-free on LE/BE for all in-tree modules. Overall this results in much simpler code, and all without the overhead of an IDR. Due to the deterministic nature, modules can now be unloaded, the congestion control algorithm for a specific but unloaded key will fall back to the default one, and on module reload time it will switch back to the expected algorithm transparently. Joint work with Florian Westphal. Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: Daniel Borkmann <dborkman@redhat.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:46 +08:00
char *tcp_ca_get_name_by_key(u32 key, char *buffer);
#else
static inline char *tcp_ca_get_name_by_key(u32 key, char *buffer)
{
return NULL;
}
#endif
net: tcp: add key management to congestion control This patch adds necessary infrastructure to the congestion control framework for later per route congestion control support. For a per route congestion control possibility, our aim is to store a unique u32 key identifier into dst metrics, which can then be mapped into a tcp_congestion_ops struct. We argue that having a RTAX key entry is the most simple, generic and easy way to manage, and also keeps the memory footprint of dst entries lower on 64 bit than with storing a pointer directly, for example. Having a unique key id also allows for decoupling actual TCP congestion control module management from the FIB layer, i.e. we don't have to care about expensive module refcounting inside the FIB at this point. We first thought of using an IDR store for the realization, which takes over dynamic assignment of unused key space and also performs the key to pointer mapping in RCU. While doing so, we stumbled upon the issue that due to the nature of dynamic key distribution, it just so happens, arguably in very rare occasions, that excessive module loads and unloads can lead to a possible reuse of previously used key space. Thus, previously stale keys in the dst metric are now being reassigned to a different congestion control algorithm, which might lead to unexpected behaviour. One way to resolve this would have been to walk FIBs on the actually rare occasion of a module unload and reset the metric keys for each FIB in each netns, but that's just very costly. Therefore, we argue a better solution is to reuse the unique congestion control algorithm name member and map that into u32 key space through jhash. For that, we split the flags attribute (as it currently uses 2 bits only anyway) into two u32 attributes, flags and key, so that we can keep the cacheline boundary of 2 cachelines on x86_64 and cache the precalculated key at registration time for the fast path. On average we might expect 2 - 4 modules being loaded worst case perhaps 15, so a key collision possibility is extremely low, and guaranteed collision-free on LE/BE for all in-tree modules. Overall this results in much simpler code, and all without the overhead of an IDR. Due to the deterministic nature, modules can now be unloaded, the congestion control algorithm for a specific but unloaded key will fall back to the default one, and on module reload time it will switch back to the expected algorithm transparently. Joint work with Florian Westphal. Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: Daniel Borkmann <dborkman@redhat.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-01-06 06:57:46 +08:00
static inline bool tcp_ca_needs_ecn(const struct sock *sk)
{
const struct inet_connection_sock *icsk = inet_csk(sk);
return icsk->icsk_ca_ops->flags & TCP_CONG_NEEDS_ECN;
}
static inline void tcp_set_ca_state(struct sock *sk, const u8 ca_state)
{
struct inet_connection_sock *icsk = inet_csk(sk);
if (icsk->icsk_ca_ops->set_state)
icsk->icsk_ca_ops->set_state(sk, ca_state);
icsk->icsk_ca_state = ca_state;
}
static inline void tcp_ca_event(struct sock *sk, const enum tcp_ca_event event)
{
const struct inet_connection_sock *icsk = inet_csk(sk);
if (icsk->icsk_ca_ops->cwnd_event)
icsk->icsk_ca_ops->cwnd_event(sk, event);
}
tcp: track data delivery rate for a TCP connection This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
/* From tcp_rate.c */
void tcp_rate_skb_sent(struct sock *sk, struct sk_buff *skb);
void tcp_rate_skb_delivered(struct sock *sk, struct sk_buff *skb,
struct rate_sample *rs);
void tcp_rate_gen(struct sock *sk, u32 delivered, u32 lost,
bool is_sack_reneg, struct rate_sample *rs);
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 11:39:15 +08:00
void tcp_rate_check_app_limited(struct sock *sk);
tcp: track data delivery rate for a TCP connection This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: Van Jacobson <vanj@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Nandita Dukkipati <nanditad@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-20 11:39:14 +08:00
/* These functions determine how the current flow behaves in respect of SACK
* handling. SACK is negotiated with the peer, and therefore it can vary
* between different flows.
*
* tcp_is_sack - SACK enabled
* tcp_is_reno - No SACK
*/
static inline int tcp_is_sack(const struct tcp_sock *tp)
{
return likely(tp->rx_opt.sack_ok);
}
static inline bool tcp_is_reno(const struct tcp_sock *tp)
{
return !tcp_is_sack(tp);
}
static inline unsigned int tcp_left_out(const struct tcp_sock *tp)
{
return tp->sacked_out + tp->lost_out;
}
/* This determines how many packets are "in the network" to the best
* of our knowledge. In many cases it is conservative, but where
* detailed information is available from the receiver (via SACK
* blocks etc.) we can make more aggressive calculations.
*
* Use this for decisions involving congestion control, use just
* tp->packets_out to determine if the send queue is empty or not.
*
* Read this equation as:
*
* "Packets sent once on transmission queue" MINUS
* "Packets left network, but not honestly ACKed yet" PLUS
* "Packets fast retransmitted"
*/
static inline unsigned int tcp_packets_in_flight(const struct tcp_sock *tp)
{
return tp->packets_out - tcp_left_out(tp) + tp->retrans_out;
}
#define TCP_INFINITE_SSTHRESH 0x7fffffff
static inline bool tcp_in_slow_start(const struct tcp_sock *tp)
{
return tp->snd_cwnd < tp->snd_ssthresh;
}
static inline bool tcp_in_initial_slowstart(const struct tcp_sock *tp)
{
return tp->snd_ssthresh >= TCP_INFINITE_SSTHRESH;
}
static inline bool tcp_in_cwnd_reduction(const struct sock *sk)
{
return (TCPF_CA_CWR | TCPF_CA_Recovery) &
(1 << inet_csk(sk)->icsk_ca_state);
}
/* If cwnd > ssthresh, we may raise ssthresh to be half-way to cwnd.
* The exception is cwnd reduction phase, when cwnd is decreasing towards
* ssthresh.
*/
static inline __u32 tcp_current_ssthresh(const struct sock *sk)
{
const struct tcp_sock *tp = tcp_sk(sk);
if (tcp_in_cwnd_reduction(sk))
return tp->snd_ssthresh;
else
return max(tp->snd_ssthresh,
((tp->snd_cwnd >> 1) +
(tp->snd_cwnd >> 2)));
}
/* Use define here intentionally to get WARN_ON location shown at the caller */
#define tcp_verify_left_out(tp) WARN_ON(tcp_left_out(tp) > tp->packets_out)
void tcp_enter_cwr(struct sock *sk);
__u32 tcp_init_cwnd(const struct tcp_sock *tp, const struct dst_entry *dst);
/* The maximum number of MSS of available cwnd for which TSO defers
* sending if not using sysctl_tcp_tso_win_divisor.
*/
static inline __u32 tcp_max_tso_deferred_mss(const struct tcp_sock *tp)
{
return 3;
}
/* Returns end sequence number of the receiver's advertised window */
static inline u32 tcp_wnd_end(const struct tcp_sock *tp)
{
return tp->snd_una + tp->snd_wnd;
}
tcp: fix cwnd limited checking to improve congestion control Yuchung discovered tcp_is_cwnd_limited() was returning false in slow start phase even if the application filled the socket write queue. All congestion modules take into account tcp_is_cwnd_limited() before increasing cwnd, so this behavior limits slow start from probing the bandwidth at full speed. The problem is that even if write queue is full (aka we are _not_ application limited), cwnd can be under utilized if TSO should auto defer or TCP Small queues decided to hold packets. So the in_flight can be kept to smaller value, and we can get to the point tcp_is_cwnd_limited() returns false. With TCP Small Queues and FQ/pacing, this issue is more visible. We fix this by having tcp_cwnd_validate(), which is supposed to track such things, take into account unsent_segs, the number of segs that we are not sending at the moment due to TSO or TSQ, but intend to send real soon. Then when we are cwnd-limited, remember this fact while we are processing the window of ACKs that comes back. For example, suppose we have a brand new connection with cwnd=10; we are in slow start, and we send a flight of 9 packets. By the time we have received ACKs for all 9 packets we want our cwnd to be 18. We implement this by setting tp->lsnd_pending to 9, and considering ourselves to be cwnd-limited while cwnd is less than twice tp->lsnd_pending (2*9 -> 18). This makes tcp_is_cwnd_limited() more understandable, by removing the GSO/TSO kludge, that tried to work around the issue. Note the in_flight parameter can be removed in a followup cleanup patch. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-05-01 02:58:13 +08:00
/* We follow the spirit of RFC2861 to validate cwnd but implement a more
* flexible approach. The RFC suggests cwnd should not be raised unless
* it was fully used previously. And that's exactly what we do in
* congestion avoidance mode. But in slow start we allow cwnd to grow
* as long as the application has used half the cwnd.
tcp: fix cwnd limited checking to improve congestion control Yuchung discovered tcp_is_cwnd_limited() was returning false in slow start phase even if the application filled the socket write queue. All congestion modules take into account tcp_is_cwnd_limited() before increasing cwnd, so this behavior limits slow start from probing the bandwidth at full speed. The problem is that even if write queue is full (aka we are _not_ application limited), cwnd can be under utilized if TSO should auto defer or TCP Small queues decided to hold packets. So the in_flight can be kept to smaller value, and we can get to the point tcp_is_cwnd_limited() returns false. With TCP Small Queues and FQ/pacing, this issue is more visible. We fix this by having tcp_cwnd_validate(), which is supposed to track such things, take into account unsent_segs, the number of segs that we are not sending at the moment due to TSO or TSQ, but intend to send real soon. Then when we are cwnd-limited, remember this fact while we are processing the window of ACKs that comes back. For example, suppose we have a brand new connection with cwnd=10; we are in slow start, and we send a flight of 9 packets. By the time we have received ACKs for all 9 packets we want our cwnd to be 18. We implement this by setting tp->lsnd_pending to 9, and considering ourselves to be cwnd-limited while cwnd is less than twice tp->lsnd_pending (2*9 -> 18). This makes tcp_is_cwnd_limited() more understandable, by removing the GSO/TSO kludge, that tried to work around the issue. Note the in_flight parameter can be removed in a followup cleanup patch. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-05-01 02:58:13 +08:00
* Example :
* cwnd is 10 (IW10), but application sends 9 frames.
* We allow cwnd to reach 18 when all frames are ACKed.
* This check is safe because it's as aggressive as slow start which already
* risks 100% overshoot. The advantage is that we discourage application to
* either send more filler packets or data to artificially blow up the cwnd
* usage, and allow application-limited process to probe bw more aggressively.
*/
static inline bool tcp_is_cwnd_limited(const struct sock *sk)
tcp: fix cwnd limited checking to improve congestion control Yuchung discovered tcp_is_cwnd_limited() was returning false in slow start phase even if the application filled the socket write queue. All congestion modules take into account tcp_is_cwnd_limited() before increasing cwnd, so this behavior limits slow start from probing the bandwidth at full speed. The problem is that even if write queue is full (aka we are _not_ application limited), cwnd can be under utilized if TSO should auto defer or TCP Small queues decided to hold packets. So the in_flight can be kept to smaller value, and we can get to the point tcp_is_cwnd_limited() returns false. With TCP Small Queues and FQ/pacing, this issue is more visible. We fix this by having tcp_cwnd_validate(), which is supposed to track such things, take into account unsent_segs, the number of segs that we are not sending at the moment due to TSO or TSQ, but intend to send real soon. Then when we are cwnd-limited, remember this fact while we are processing the window of ACKs that comes back. For example, suppose we have a brand new connection with cwnd=10; we are in slow start, and we send a flight of 9 packets. By the time we have received ACKs for all 9 packets we want our cwnd to be 18. We implement this by setting tp->lsnd_pending to 9, and considering ourselves to be cwnd-limited while cwnd is less than twice tp->lsnd_pending (2*9 -> 18). This makes tcp_is_cwnd_limited() more understandable, by removing the GSO/TSO kludge, that tried to work around the issue. Note the in_flight parameter can be removed in a followup cleanup patch. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-05-01 02:58:13 +08:00
{
const struct tcp_sock *tp = tcp_sk(sk);
/* If in slow start, ensure cwnd grows to twice what was ACKed. */
if (tcp_in_slow_start(tp))
return tp->snd_cwnd < 2 * tp->max_packets_out;
return tp->is_cwnd_limited;
tcp: fix cwnd limited checking to improve congestion control Yuchung discovered tcp_is_cwnd_limited() was returning false in slow start phase even if the application filled the socket write queue. All congestion modules take into account tcp_is_cwnd_limited() before increasing cwnd, so this behavior limits slow start from probing the bandwidth at full speed. The problem is that even if write queue is full (aka we are _not_ application limited), cwnd can be under utilized if TSO should auto defer or TCP Small queues decided to hold packets. So the in_flight can be kept to smaller value, and we can get to the point tcp_is_cwnd_limited() returns false. With TCP Small Queues and FQ/pacing, this issue is more visible. We fix this by having tcp_cwnd_validate(), which is supposed to track such things, take into account unsent_segs, the number of segs that we are not sending at the moment due to TSO or TSQ, but intend to send real soon. Then when we are cwnd-limited, remember this fact while we are processing the window of ACKs that comes back. For example, suppose we have a brand new connection with cwnd=10; we are in slow start, and we send a flight of 9 packets. By the time we have received ACKs for all 9 packets we want our cwnd to be 18. We implement this by setting tp->lsnd_pending to 9, and considering ourselves to be cwnd-limited while cwnd is less than twice tp->lsnd_pending (2*9 -> 18). This makes tcp_is_cwnd_limited() more understandable, by removing the GSO/TSO kludge, that tried to work around the issue. Note the in_flight parameter can be removed in a followup cleanup patch. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2014-05-01 02:58:13 +08:00
}
/* BBR congestion control needs pacing.
* Same remark for SO_MAX_PACING_RATE.
* sch_fq packet scheduler is efficiently handling pacing,
* but is not always installed/used.
* Return true if TCP stack should pace packets itself.
*/
static inline bool tcp_needs_internal_pacing(const struct sock *sk)
{
return smp_load_acquire(&sk->sk_pacing_status) == SK_PACING_NEEDED;
}
/* Estimates in how many jiffies next packet for this flow can be sent.
* Scheduling a retransmit timer too early would be silly.
*/
static inline unsigned long tcp_pacing_delay(const struct sock *sk)
{
s64 delay = tcp_sk(sk)->tcp_wstamp_ns - tcp_sk(sk)->tcp_clock_cache;
return delay > 0 ? nsecs_to_jiffies(delay) : 0;
}
static inline void tcp_reset_xmit_timer(struct sock *sk,
const int what,
unsigned long when,
const unsigned long max_when)
{
inet_csk_reset_xmit_timer(sk, what, when + tcp_pacing_delay(sk),
max_when);
}
/* Something is really bad, we could not queue an additional packet,
* because qdisc is full or receiver sent a 0 window, or we are paced.
* We do not want to add fuel to the fire, or abort too early,
* so make sure the timer we arm now is at least 200ms in the future,
* regardless of current icsk_rto value (as it could be ~2ms)
*/
static inline unsigned long tcp_probe0_base(const struct sock *sk)
{
return max_t(unsigned long, inet_csk(sk)->icsk_rto, TCP_RTO_MIN);
}
[TCP]: Sed magic converts func(sk, tp, ...) -> func(sk, ...) This is (mostly) automated change using magic: sed -e '/struct sock \*sk/ N' -e '/struct sock \*sk/ N' -e '/struct sock \*sk/ N' -e '/struct sock \*sk/ N' -e 's|struct sock \*sk,[\n\t ]*struct tcp_sock \*tp\([^{]*\n{\n\)| struct sock \*sk\1\tstruct tcp_sock *tp = tcp_sk(sk);\n|g' -e 's|struct sock \*sk, struct tcp_sock \*tp| struct sock \*sk|g' -e 's|sk, tp\([^-]\)|sk\1|g' Fixed four unused variable (tp) warnings that were introduced. In addition, manually added newlines after local variables and tweaked function arguments positioning. $ gcc --version gcc (GCC) 4.1.1 20060525 (Red Hat 4.1.1-1) ... $ codiff -fV built-in.o.old built-in.o.new net/ipv4/route.c: rt_cache_flush | +14 1 function changed, 14 bytes added net/ipv4/tcp.c: tcp_setsockopt | -5 tcp_sendpage | -25 tcp_sendmsg | -16 3 functions changed, 46 bytes removed net/ipv4/tcp_input.c: tcp_try_undo_recovery | +3 tcp_try_undo_dsack | +2 tcp_mark_head_lost | -12 tcp_ack | -15 tcp_event_data_recv | -32 tcp_rcv_state_process | -10 tcp_rcv_established | +1 7 functions changed, 6 bytes added, 69 bytes removed, diff: -63 net/ipv4/tcp_output.c: update_send_head | -9 tcp_transmit_skb | +19 tcp_cwnd_validate | +1 tcp_write_wakeup | -17 __tcp_push_pending_frames | -25 tcp_push_one | -8 tcp_send_fin | -4 7 functions changed, 20 bytes added, 63 bytes removed, diff: -43 built-in.o.new: 18 functions changed, 40 bytes added, 178 bytes removed, diff: -138 Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Signed-off-by: David S. Miller <davem@davemloft.net>
2007-04-21 13:18:02 +08:00
/* Variant of inet_csk_rto_backoff() used for zero window probes */
static inline unsigned long tcp_probe0_when(const struct sock *sk,
unsigned long max_when)
{
u64 when = (u64)tcp_probe0_base(sk) << inet_csk(sk)->icsk_backoff;
return (unsigned long)min_t(u64, when, max_when);
}
static inline void tcp_check_probe_timer(struct sock *sk)
{
if (!tcp_sk(sk)->packets_out && !inet_csk(sk)->icsk_pending)
tcp_reset_xmit_timer(sk, ICSK_TIME_PROBE0,
tcp_probe0_base(sk), TCP_RTO_MAX);
}
static inline void tcp_init_wl(struct tcp_sock *tp, u32 seq)
{
tp->snd_wl1 = seq;
}
static inline void tcp_update_wl(struct tcp_sock *tp, u32 seq)
{
tp->snd_wl1 = seq;
}
/*
* Calculate(/check) TCP checksum
*/
static inline __sum16 tcp_v4_check(int len, __be32 saddr,
__be32 daddr, __wsum base)
{
return csum_tcpudp_magic(saddr, daddr, len, IPPROTO_TCP, base);
}
static inline bool tcp_checksum_complete(struct sk_buff *skb)
{
return !skb_csum_unnecessary(skb) &&
__skb_checksum_complete(skb);
}
bool tcp_add_backlog(struct sock *sk, struct sk_buff *skb);
int tcp_filter(struct sock *sk, struct sk_buff *skb);
void tcp_set_state(struct sock *sk, int state);
void tcp_done(struct sock *sk);
int tcp_abort(struct sock *sk, int err);
static inline void tcp_sack_reset(struct tcp_options_received *rx_opt)
{
rx_opt->dsack = 0;
rx_opt->num_sacks = 0;
}
tcp: fix slow start after idle vs TSO/GSO slow start after idle might reduce cwnd, but we perform this after first packet was cooked and sent. With TSO/GSO, it means that we might send a full TSO packet even if cwnd should have been reduced to IW10. Moving the SSAI check in skb_entail() makes sense, because we slightly reduce number of times this check is done, especially for large send() and TCP Small queue callbacks from softirq context. As Neal pointed out, we also need to perform the check if/when receive window opens. Tested: Following packetdrill test demonstrates the problem // Test of slow start after idle `sysctl -q net.ipv4.tcp_slow_start_after_idle=1` 0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 +0 < S 0:0(0) win 65535 <mss 1000,sackOK,nop,nop,nop,wscale 7> +0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6> +.100 < . 1:1(0) ack 1 win 511 +0 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_SOCKET, SO_SNDBUF, [200000], 4) = 0 +0 write(4, ..., 26000) = 26000 +0 > . 1:5001(5000) ack 1 +0 > . 5001:10001(5000) ack 1 +0 %{ assert tcpi_snd_cwnd == 10 }% +.100 < . 1:1(0) ack 10001 win 511 +0 %{ assert tcpi_snd_cwnd == 20, tcpi_snd_cwnd }% +0 > . 10001:20001(10000) ack 1 +0 > P. 20001:26001(6000) ack 1 +.100 < . 1:1(0) ack 26001 win 511 +0 %{ assert tcpi_snd_cwnd == 36, tcpi_snd_cwnd }% +4 write(4, ..., 20000) = 20000 // If slow start after idle works properly, we should send 5 MSS here (cwnd/2) +0 > . 26001:31001(5000) ack 1 +0 %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }% +0 > . 31001:36001(5000) ack 1 Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-08-22 03:30:00 +08:00
void tcp_cwnd_restart(struct sock *sk, s32 delta);
static inline void tcp_slow_start_after_idle_check(struct sock *sk)
{
const struct tcp_congestion_ops *ca_ops = inet_csk(sk)->icsk_ca_ops;
tcp: fix slow start after idle vs TSO/GSO slow start after idle might reduce cwnd, but we perform this after first packet was cooked and sent. With TSO/GSO, it means that we might send a full TSO packet even if cwnd should have been reduced to IW10. Moving the SSAI check in skb_entail() makes sense, because we slightly reduce number of times this check is done, especially for large send() and TCP Small queue callbacks from softirq context. As Neal pointed out, we also need to perform the check if/when receive window opens. Tested: Following packetdrill test demonstrates the problem // Test of slow start after idle `sysctl -q net.ipv4.tcp_slow_start_after_idle=1` 0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 +0 < S 0:0(0) win 65535 <mss 1000,sackOK,nop,nop,nop,wscale 7> +0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6> +.100 < . 1:1(0) ack 1 win 511 +0 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_SOCKET, SO_SNDBUF, [200000], 4) = 0 +0 write(4, ..., 26000) = 26000 +0 > . 1:5001(5000) ack 1 +0 > . 5001:10001(5000) ack 1 +0 %{ assert tcpi_snd_cwnd == 10 }% +.100 < . 1:1(0) ack 10001 win 511 +0 %{ assert tcpi_snd_cwnd == 20, tcpi_snd_cwnd }% +0 > . 10001:20001(10000) ack 1 +0 > P. 20001:26001(6000) ack 1 +.100 < . 1:1(0) ack 26001 win 511 +0 %{ assert tcpi_snd_cwnd == 36, tcpi_snd_cwnd }% +4 write(4, ..., 20000) = 20000 // If slow start after idle works properly, we should send 5 MSS here (cwnd/2) +0 > . 26001:31001(5000) ack 1 +0 %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }% +0 > . 31001:36001(5000) ack 1 Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-08-22 03:30:00 +08:00
struct tcp_sock *tp = tcp_sk(sk);
s32 delta;
if (!sock_net(sk)->ipv4.sysctl_tcp_slow_start_after_idle || tp->packets_out ||
ca_ops->cong_control)
tcp: fix slow start after idle vs TSO/GSO slow start after idle might reduce cwnd, but we perform this after first packet was cooked and sent. With TSO/GSO, it means that we might send a full TSO packet even if cwnd should have been reduced to IW10. Moving the SSAI check in skb_entail() makes sense, because we slightly reduce number of times this check is done, especially for large send() and TCP Small queue callbacks from softirq context. As Neal pointed out, we also need to perform the check if/when receive window opens. Tested: Following packetdrill test demonstrates the problem // Test of slow start after idle `sysctl -q net.ipv4.tcp_slow_start_after_idle=1` 0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 +0 < S 0:0(0) win 65535 <mss 1000,sackOK,nop,nop,nop,wscale 7> +0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6> +.100 < . 1:1(0) ack 1 win 511 +0 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_SOCKET, SO_SNDBUF, [200000], 4) = 0 +0 write(4, ..., 26000) = 26000 +0 > . 1:5001(5000) ack 1 +0 > . 5001:10001(5000) ack 1 +0 %{ assert tcpi_snd_cwnd == 10 }% +.100 < . 1:1(0) ack 10001 win 511 +0 %{ assert tcpi_snd_cwnd == 20, tcpi_snd_cwnd }% +0 > . 10001:20001(10000) ack 1 +0 > P. 20001:26001(6000) ack 1 +.100 < . 1:1(0) ack 26001 win 511 +0 %{ assert tcpi_snd_cwnd == 36, tcpi_snd_cwnd }% +4 write(4, ..., 20000) = 20000 // If slow start after idle works properly, we should send 5 MSS here (cwnd/2) +0 > . 26001:31001(5000) ack 1 +0 %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }% +0 > . 31001:36001(5000) ack 1 Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-08-22 03:30:00 +08:00
return;
delta = tcp_jiffies32 - tp->lsndtime;
tcp: fix slow start after idle vs TSO/GSO slow start after idle might reduce cwnd, but we perform this after first packet was cooked and sent. With TSO/GSO, it means that we might send a full TSO packet even if cwnd should have been reduced to IW10. Moving the SSAI check in skb_entail() makes sense, because we slightly reduce number of times this check is done, especially for large send() and TCP Small queue callbacks from softirq context. As Neal pointed out, we also need to perform the check if/when receive window opens. Tested: Following packetdrill test demonstrates the problem // Test of slow start after idle `sysctl -q net.ipv4.tcp_slow_start_after_idle=1` 0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 +0 < S 0:0(0) win 65535 <mss 1000,sackOK,nop,nop,nop,wscale 7> +0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6> +.100 < . 1:1(0) ack 1 win 511 +0 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_SOCKET, SO_SNDBUF, [200000], 4) = 0 +0 write(4, ..., 26000) = 26000 +0 > . 1:5001(5000) ack 1 +0 > . 5001:10001(5000) ack 1 +0 %{ assert tcpi_snd_cwnd == 10 }% +.100 < . 1:1(0) ack 10001 win 511 +0 %{ assert tcpi_snd_cwnd == 20, tcpi_snd_cwnd }% +0 > . 10001:20001(10000) ack 1 +0 > P. 20001:26001(6000) ack 1 +.100 < . 1:1(0) ack 26001 win 511 +0 %{ assert tcpi_snd_cwnd == 36, tcpi_snd_cwnd }% +4 write(4, ..., 20000) = 20000 // If slow start after idle works properly, we should send 5 MSS here (cwnd/2) +0 > . 26001:31001(5000) ack 1 +0 %{ assert tcpi_snd_cwnd == 10, tcpi_snd_cwnd }% +0 > . 31001:36001(5000) ack 1 Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-08-22 03:30:00 +08:00
if (delta > inet_csk(sk)->icsk_rto)
tcp_cwnd_restart(sk, delta);
}
/* Determine a window scaling and initial window to offer. */
void tcp_select_initial_window(const struct sock *sk, int __space,
__u32 mss, __u32 *rcv_wnd,
__u32 *window_clamp, int wscale_ok,
__u8 *rcv_wscale, __u32 init_rcv_wnd);
static inline int tcp_win_from_space(const struct sock *sk, int space)
{
int tcp_adv_win_scale = sock_net(sk)->ipv4.sysctl_tcp_adv_win_scale;
return tcp_adv_win_scale <= 0 ?
(space>>(-tcp_adv_win_scale)) :
space - (space>>tcp_adv_win_scale);
}
/* Note: caller must be prepared to deal with negative returns */
static inline int tcp_space(const struct sock *sk)
{
return tcp_win_from_space(sk, READ_ONCE(sk->sk_rcvbuf) -
net: silence KCSAN warnings about sk->sk_backlog.len reads sk->sk_backlog.len can be written by BH handlers, and read from process contexts in a lockless way. Note the write side should also use WRITE_ONCE() or a variant. We need some agreement about the best way to do this. syzbot reported : BUG: KCSAN: data-race in tcp_add_backlog / tcp_grow_window.isra.0 write to 0xffff88812665f32c of 4 bytes by interrupt on cpu 1: sk_add_backlog include/net/sock.h:934 [inline] tcp_add_backlog+0x4a0/0xcc0 net/ipv4/tcp_ipv4.c:1737 tcp_v4_rcv+0x1aba/0x1bf0 net/ipv4/tcp_ipv4.c:1925 ip_protocol_deliver_rcu+0x51/0x470 net/ipv4/ip_input.c:204 ip_local_deliver_finish+0x110/0x140 net/ipv4/ip_input.c:231 NF_HOOK include/linux/netfilter.h:305 [inline] NF_HOOK include/linux/netfilter.h:299 [inline] ip_local_deliver+0x133/0x210 net/ipv4/ip_input.c:252 dst_input include/net/dst.h:442 [inline] ip_rcv_finish+0x121/0x160 net/ipv4/ip_input.c:413 NF_HOOK include/linux/netfilter.h:305 [inline] NF_HOOK include/linux/netfilter.h:299 [inline] ip_rcv+0x18f/0x1a0 net/ipv4/ip_input.c:523 __netif_receive_skb_one_core+0xa7/0xe0 net/core/dev.c:5004 __netif_receive_skb+0x37/0xf0 net/core/dev.c:5118 netif_receive_skb_internal+0x59/0x190 net/core/dev.c:5208 napi_skb_finish net/core/dev.c:5671 [inline] napi_gro_receive+0x28f/0x330 net/core/dev.c:5704 receive_buf+0x284/0x30b0 drivers/net/virtio_net.c:1061 virtnet_receive drivers/net/virtio_net.c:1323 [inline] virtnet_poll+0x436/0x7d0 drivers/net/virtio_net.c:1428 napi_poll net/core/dev.c:6352 [inline] net_rx_action+0x3ae/0xa50 net/core/dev.c:6418 read to 0xffff88812665f32c of 4 bytes by task 7292 on cpu 0: tcp_space include/net/tcp.h:1373 [inline] tcp_grow_window.isra.0+0x6b/0x480 net/ipv4/tcp_input.c:413 tcp_event_data_recv+0x68f/0x990 net/ipv4/tcp_input.c:717 tcp_rcv_established+0xbfe/0xf50 net/ipv4/tcp_input.c:5618 tcp_v4_do_rcv+0x381/0x4e0 net/ipv4/tcp_ipv4.c:1542 sk_backlog_rcv include/net/sock.h:945 [inline] __release_sock+0x135/0x1e0 net/core/sock.c:2427 release_sock+0x61/0x160 net/core/sock.c:2943 tcp_recvmsg+0x63b/0x1a30 net/ipv4/tcp.c:2181 inet_recvmsg+0xbb/0x250 net/ipv4/af_inet.c:838 sock_recvmsg_nosec net/socket.c:871 [inline] sock_recvmsg net/socket.c:889 [inline] sock_recvmsg+0x92/0xb0 net/socket.c:885 sock_read_iter+0x15f/0x1e0 net/socket.c:967 call_read_iter include/linux/fs.h:1864 [inline] new_sync_read+0x389/0x4f0 fs/read_write.c:414 __vfs_read+0xb1/0xc0 fs/read_write.c:427 vfs_read fs/read_write.c:461 [inline] vfs_read+0x143/0x2c0 fs/read_write.c:446 Reported by Kernel Concurrency Sanitizer on: CPU: 0 PID: 7292 Comm: syz-fuzzer Not tainted 5.3.0+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Signed-off-by: Eric Dumazet <edumazet@google.com> Reported-by: syzbot <syzkaller@googlegroups.com> Signed-off-by: Jakub Kicinski <jakub.kicinski@netronome.com>
2019-10-10 06:41:03 +08:00
READ_ONCE(sk->sk_backlog.len) -
atomic_read(&sk->sk_rmem_alloc));
}
static inline int tcp_full_space(const struct sock *sk)
{
return tcp_win_from_space(sk, READ_ONCE(sk->sk_rcvbuf));
}
void tcp_cleanup_rbuf(struct sock *sk, int copied);
/* We provision sk_rcvbuf around 200% of sk_rcvlowat.
* If 87.5 % (7/8) of the space has been consumed, we want to override
* SO_RCVLOWAT constraint, since we are receiving skbs with too small
* len/truesize ratio.
*/
static inline bool tcp_rmem_pressure(const struct sock *sk)
{
int rcvbuf = READ_ONCE(sk->sk_rcvbuf);
int threshold = rcvbuf - (rcvbuf >> 3);
return atomic_read(&sk->sk_rmem_alloc) > threshold;
}
extern void tcp_openreq_init_rwin(struct request_sock *req,
const struct sock *sk_listener,
const struct dst_entry *dst);
void tcp_enter_memory_pressure(struct sock *sk);
void tcp_leave_memory_pressure(struct sock *sk);
static inline int keepalive_intvl_when(const struct tcp_sock *tp)
{
struct net *net = sock_net((struct sock *)tp);
return tp->keepalive_intvl ? : net->ipv4.sysctl_tcp_keepalive_intvl;
}
static inline int keepalive_time_when(const struct tcp_sock *tp)
{
struct net *net = sock_net((struct sock *)tp);
return tp->keepalive_time ? : net->ipv4.sysctl_tcp_keepalive_time;
}
static inline int keepalive_probes(const struct tcp_sock *tp)
{
struct net *net = sock_net((struct sock *)tp);
return tp->keepalive_probes ? : net->ipv4.sysctl_tcp_keepalive_probes;
}
static inline u32 keepalive_time_elapsed(const struct tcp_sock *tp)
{
const struct inet_connection_sock *icsk = &tp->inet_conn;
return min_t(u32, tcp_jiffies32 - icsk->icsk_ack.lrcvtime,
tcp_jiffies32 - tp->rcv_tstamp);
}
static inline int tcp_fin_time(const struct sock *sk)
{
int fin_timeout = tcp_sk(sk)->linger2 ? : sock_net(sk)->ipv4.sysctl_tcp_fin_timeout;
const int rto = inet_csk(sk)->icsk_rto;
if (fin_timeout < (rto << 2) - (rto >> 1))
fin_timeout = (rto << 2) - (rto >> 1);
return fin_timeout;
}
static inline bool tcp_paws_check(const struct tcp_options_received *rx_opt,
int paws_win)
{
if ((s32)(rx_opt->ts_recent - rx_opt->rcv_tsval) <= paws_win)
return true;
if (unlikely(!time_before32(ktime_get_seconds(),
rx_opt->ts_recent_stamp + TCP_PAWS_24DAYS)))
return true;
/*
* Some OSes send SYN and SYNACK messages with tsval=0 tsecr=0,
* then following tcp messages have valid values. Ignore 0 value,
* or else 'negative' tsval might forbid us to accept their packets.
*/
if (!rx_opt->ts_recent)
return true;
return false;
}
static inline bool tcp_paws_reject(const struct tcp_options_received *rx_opt,
int rst)
{
if (tcp_paws_check(rx_opt, 0))
return false;
/* RST segments are not recommended to carry timestamp,
and, if they do, it is recommended to ignore PAWS because
"their cleanup function should take precedence over timestamps."
Certainly, it is mistake. It is necessary to understand the reasons
of this constraint to relax it: if peer reboots, clock may go
out-of-sync and half-open connections will not be reset.
Actually, the problem would be not existing if all
the implementations followed draft about maintaining clock
via reboots. Linux-2.2 DOES NOT!
However, we can relax time bounds for RST segments to MSL.
*/
if (rst && !time_before32(ktime_get_seconds(),
rx_opt->ts_recent_stamp + TCP_PAWS_MSL))
return false;
return true;
}
bool tcp_oow_rate_limited(struct net *net, const struct sk_buff *skb,
int mib_idx, u32 *last_oow_ack_time);
tcp: helpers to mitigate ACK loops by rate-limiting out-of-window dupacks Helpers for mitigating ACK loops by rate-limiting dupacks sent in response to incoming out-of-window packets. This patch includes: - rate-limiting logic - sysctl to control how often we allow dupacks to out-of-window packets - SNMP counter for cases where we rate-limited our dupack sending The rate-limiting logic in this patch decides to not send dupacks in response to out-of-window segments if (a) they are SYNs or pure ACKs and (b) the remote endpoint is sending them faster than the configured rate limit. We rate-limit our responses rather than blocking them entirely or resetting the connection, because legitimate connections can rely on dupacks in response to some out-of-window segments. For example, zero window probes are typically sent with a sequence number that is below the current window, and ZWPs thus expect to thus elicit a dupack in response. We allow dupacks in response to TCP segments with data, because these may be spurious retransmissions for which the remote endpoint wants to receive DSACKs. This is safe because segments with data can't realistically be part of ACK loops, which by their nature consist of each side sending pure/data-less ACKs to each other. The dupack interval is controlled by a new sysctl knob, tcp_invalid_ratelimit, given in milliseconds, in case an administrator needs to dial this upward in the face of a high-rate DoS attack. The name and units are chosen to be analogous to the existing analogous knob for ICMP, icmp_ratelimit. The default value for tcp_invalid_ratelimit is 500ms, which allows at most one such dupack per 500ms. This is chosen to be 2x faster than the 1-second minimum RTO interval allowed by RFC 6298 (section 2, rule 2.4). We allow the extra 2x factor because network delay variations can cause packets sent at 1 second intervals to be compressed and arrive much closer. Reported-by: Avery Fay <avery@mixpanel.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2015-02-07 05:04:38 +08:00
static inline void tcp_mib_init(struct net *net)
{
/* See RFC 2012 */
TCP_ADD_STATS(net, TCP_MIB_RTOALGORITHM, 1);
TCP_ADD_STATS(net, TCP_MIB_RTOMIN, TCP_RTO_MIN*1000/HZ);
TCP_ADD_STATS(net, TCP_MIB_RTOMAX, TCP_RTO_MAX*1000/HZ);
TCP_ADD_STATS(net, TCP_MIB_MAXCONN, -1);
}
/* from STCP */
static inline void tcp_clear_retrans_hints_partial(struct tcp_sock *tp)
{
tp->lost_skb_hint = NULL;
}
static inline void tcp_clear_all_retrans_hints(struct tcp_sock *tp)
{
tcp_clear_retrans_hints_partial(tp);
tp->retransmit_skb_hint = NULL;
}
union tcp_md5_addr {
struct in_addr a4;
#if IS_ENABLED(CONFIG_IPV6)
struct in6_addr a6;
#endif
};
/* - key database */
struct tcp_md5sig_key {
struct hlist_node node;
u8 keylen;
u8 family; /* AF_INET or AF_INET6 */
u8 prefixlen;
union tcp_md5_addr addr;
int l3index; /* set if key added with L3 scope */
u8 key[TCP_MD5SIG_MAXKEYLEN];
struct rcu_head rcu;
};
/* - sock block */
struct tcp_md5sig_info {
struct hlist_head head;
struct rcu_head rcu;
};
/* - pseudo header */
struct tcp4_pseudohdr {
__be32 saddr;
__be32 daddr;
__u8 pad;
__u8 protocol;
__be16 len;
};
struct tcp6_pseudohdr {
struct in6_addr saddr;
struct in6_addr daddr;
__be32 len;
__be32 protocol; /* including padding */
};
union tcp_md5sum_block {
struct tcp4_pseudohdr ip4;
#if IS_ENABLED(CONFIG_IPV6)
struct tcp6_pseudohdr ip6;
#endif
};
/* - pool: digest algorithm, hash description and scratch buffer */
struct tcp_md5sig_pool {
struct ahash_request *md5_req;
void *scratch;
};
/* - functions */
int tcp_v4_md5_hash_skb(char *md5_hash, const struct tcp_md5sig_key *key,
const struct sock *sk, const struct sk_buff *skb);
int tcp_md5_do_add(struct sock *sk, const union tcp_md5_addr *addr,
int family, u8 prefixlen, int l3index,
const u8 *newkey, u8 newkeylen, gfp_t gfp);
int tcp_md5_do_del(struct sock *sk, const union tcp_md5_addr *addr,
int family, u8 prefixlen, int l3index);
struct tcp_md5sig_key *tcp_v4_md5_lookup(const struct sock *sk,
const struct sock *addr_sk);
#ifdef CONFIG_TCP_MD5SIG
#include <linux/jump_label.h>
extern struct static_key_false tcp_md5_needed;
struct tcp_md5sig_key *__tcp_md5_do_lookup(const struct sock *sk, int l3index,
const union tcp_md5_addr *addr,
int family);
static inline struct tcp_md5sig_key *
tcp_md5_do_lookup(const struct sock *sk, int l3index,
const union tcp_md5_addr *addr, int family)
{
if (!static_branch_unlikely(&tcp_md5_needed))
return NULL;
return __tcp_md5_do_lookup(sk, l3index, addr, family);
}
#define tcp_twsk_md5_key(twsk) ((twsk)->tw_md5_key)
#else
static inline struct tcp_md5sig_key *
tcp_md5_do_lookup(const struct sock *sk, int l3index,
const union tcp_md5_addr *addr, int family)
{
return NULL;
}
#define tcp_twsk_md5_key(twsk) NULL
#endif
bool tcp_alloc_md5sig_pool(void);
struct tcp_md5sig_pool *tcp_get_md5sig_pool(void);
static inline void tcp_put_md5sig_pool(void)
{
local_bh_enable();
}
int tcp_md5_hash_skb_data(struct tcp_md5sig_pool *, const struct sk_buff *,
unsigned int header_len);
int tcp_md5_hash_key(struct tcp_md5sig_pool *hp,
const struct tcp_md5sig_key *key);
/* From tcp_fastopen.c */
void tcp_fastopen_cache_get(struct sock *sk, u16 *mss,
struct tcp_fastopen_cookie *cookie);
void tcp_fastopen_cache_set(struct sock *sk, u16 mss,
struct tcp_fastopen_cookie *cookie, bool syn_lost,
u16 try_exp);
struct tcp_fastopen_request {
/* Fast Open cookie. Size 0 means a cookie request */
struct tcp_fastopen_cookie cookie;
struct msghdr *data; /* data in MSG_FASTOPEN */
size_t size;
int copied; /* queued in tcp_connect() */
struct ubuf_info *uarg;
};
void tcp_free_fastopen_req(struct tcp_sock *tp);
void tcp_fastopen_destroy_cipher(struct sock *sk);
void tcp_fastopen_ctx_destroy(struct net *net);
int tcp_fastopen_reset_cipher(struct net *net, struct sock *sk,
void *primary_key, void *backup_key);
int tcp_fastopen_get_cipher(struct net *net, struct inet_connection_sock *icsk,
u64 *key);
void tcp_fastopen_add_skb(struct sock *sk, struct sk_buff *skb);
struct sock *tcp_try_fastopen(struct sock *sk, struct sk_buff *skb,
struct request_sock *req,
struct tcp_fastopen_cookie *foc,
const struct dst_entry *dst);
void tcp_fastopen_init_key_once(struct net *net);
bool tcp_fastopen_cookie_check(struct sock *sk, u16 *mss,
struct tcp_fastopen_cookie *cookie);
net/tcp-fastopen: Add new API support This patch adds a new socket option, TCP_FASTOPEN_CONNECT, as an alternative way to perform Fast Open on the active side (client). Prior to this patch, a client needs to replace the connect() call with sendto(MSG_FASTOPEN). This can be cumbersome for applications who want to use Fast Open: these socket operations are often done in lower layer libraries used by many other applications. Changing these libraries and/or the socket call sequences are not trivial. A more convenient approach is to perform Fast Open by simply enabling a socket option when the socket is created w/o changing other socket calls sequence: s = socket() create a new socket setsockopt(s, IPPROTO_TCP, TCP_FASTOPEN_CONNECT …); newly introduced sockopt If set, new functionality described below will be used. Return ENOTSUPP if TFO is not supported or not enabled in the kernel. connect() With cookie present, return 0 immediately. With no cookie, initiate 3WHS with TFO cookie-request option and return -1 with errno = EINPROGRESS. write()/sendmsg() With cookie present, send out SYN with data and return the number of bytes buffered. With no cookie, and 3WHS not yet completed, return -1 with errno = EINPROGRESS. No MSG_FASTOPEN flag is needed. read() Return -1 with errno = EWOULDBLOCK/EAGAIN if connect() is called but write() is not called yet. Return -1 with errno = EWOULDBLOCK/EAGAIN if connection is established but no msg is received yet. Return number of bytes read if socket is established and there is msg received. The new API simplifies life for applications that always perform a write() immediately after a successful connect(). Such applications can now take advantage of Fast Open by merely making one new setsockopt() call at the time of creating the socket. Nothing else about the application's socket call sequence needs to change. Signed-off-by: Wei Wang <weiwan@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Acked-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-01-24 02:59:22 +08:00
bool tcp_fastopen_defer_connect(struct sock *sk, int *err);
#define TCP_FASTOPEN_KEY_LENGTH sizeof(siphash_key_t)
#define TCP_FASTOPEN_KEY_MAX 2
#define TCP_FASTOPEN_KEY_BUF_LENGTH \
(TCP_FASTOPEN_KEY_LENGTH * TCP_FASTOPEN_KEY_MAX)
/* Fastopen key context */
struct tcp_fastopen_context {
siphash_key_t key[TCP_FASTOPEN_KEY_MAX];
net: ipv4: move tcp_fastopen server side code to SipHash library Using a bare block cipher in non-crypto code is almost always a bad idea, not only for security reasons (and we've seen some examples of this in the kernel in the past), but also for performance reasons. In the TCP fastopen case, we call into the bare AES block cipher one or two times (depending on whether the connection is IPv4 or IPv6). On most systems, this results in a call chain such as crypto_cipher_encrypt_one(ctx, dst, src) crypto_cipher_crt(tfm)->cit_encrypt_one(crypto_cipher_tfm(tfm), ...); aesni_encrypt kernel_fpu_begin(); aesni_enc(ctx, dst, src); // asm routine kernel_fpu_end(); It is highly unlikely that the use of special AES instructions has a benefit in this case, especially since we are doing the above twice for IPv6 connections, instead of using a transform which can process the entire input in one go. We could switch to the cbcmac(aes) shash, which would at least get rid of the duplicated overhead in *some* cases (i.e., today, only arm64 has an accelerated implementation of cbcmac(aes), while x86 will end up using the generic cbcmac template wrapping the AES-NI cipher, which basically ends up doing exactly the above). However, in the given context, it makes more sense to use a light-weight MAC algorithm that is more suitable for the purpose at hand, such as SipHash. Since the output size of SipHash already matches our chosen value for TCP_FASTOPEN_COOKIE_SIZE, and given that it accepts arbitrary input sizes, this greatly simplifies the code as well. NOTE: Server farms backing a single server IP for load balancing purposes and sharing a single fastopen key will be adversely affected by this change unless all systems in the pool receive their kernel upgrades at the same time. Signed-off-by: Ard Biesheuvel <ard.biesheuvel@linaro.org> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2019-06-17 16:09:33 +08:00
int num;
struct rcu_head rcu;
};
net/tcp_fastopen: Disable active side TFO in certain scenarios Middlebox firewall issues can potentially cause server's data being blackholed after a successful 3WHS using TFO. Following are the related reports from Apple: https://www.nanog.org/sites/default/files/Paasch_Network_Support.pdf Slide 31 identifies an issue where the client ACK to the server's data sent during a TFO'd handshake is dropped. C ---> syn-data ---> S C <--- syn/ack ----- S C (accept & write) C <---- data ------- S C ----- ACK -> X S [retry and timeout] https://www.ietf.org/proceedings/94/slides/slides-94-tcpm-13.pdf Slide 5 shows a similar situation that the server's data gets dropped after 3WHS. C ---- syn-data ---> S C <--- syn/ack ----- S C ---- ack --------> S S (accept & write) C? X <- data ------ S [retry and timeout] This is the worst failure b/c the client can not detect such behavior to mitigate the situation (such as disabling TFO). Failing to proceed, the application (e.g., SSL library) may simply timeout and retry with TFO again, and the process repeats indefinitely. The proposed solution is to disable active TFO globally under the following circumstances: 1. client side TFO socket detects out of order FIN 2. client side TFO socket receives out of order RST We disable active side TFO globally for 1hr at first. Then if it happens again, we disable it for 2h, then 4h, 8h, ... And we reset the timeout to 1hr if a client side TFO sockets not opened on loopback has successfully received data segs from server. And we examine this condition during close(). The rational behind it is that when such firewall issue happens, application running on the client should eventually close the socket as it is not able to get the data it is expecting. Or application running on the server should close the socket as it is not able to receive any response from client. In both cases, out of order FIN or RST will get received on the client given that the firewall will not block them as no data are in those frames. And we want to disable active TFO globally as it helps if the middle box is very close to the client and most of the connections are likely to fail. Also, add a debug sysctl: tcp_fastopen_blackhole_detect_timeout_sec: the initial timeout to use when firewall blackhole issue happens. This can be set and read. When setting it to 0, it means to disable the active disable logic. Signed-off-by: Wei Wang <weiwan@google.com> Acked-by: Yuchung Cheng <ycheng@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-04-21 05:45:46 +08:00
extern unsigned int sysctl_tcp_fastopen_blackhole_timeout;
void tcp_fastopen_active_disable(struct sock *sk);
net/tcp_fastopen: Disable active side TFO in certain scenarios Middlebox firewall issues can potentially cause server's data being blackholed after a successful 3WHS using TFO. Following are the related reports from Apple: https://www.nanog.org/sites/default/files/Paasch_Network_Support.pdf Slide 31 identifies an issue where the client ACK to the server's data sent during a TFO'd handshake is dropped. C ---> syn-data ---> S C <--- syn/ack ----- S C (accept & write) C <---- data ------- S C ----- ACK -> X S [retry and timeout] https://www.ietf.org/proceedings/94/slides/slides-94-tcpm-13.pdf Slide 5 shows a similar situation that the server's data gets dropped after 3WHS. C ---- syn-data ---> S C <--- syn/ack ----- S C ---- ack --------> S S (accept & write) C? X <- data ------ S [retry and timeout] This is the worst failure b/c the client can not detect such behavior to mitigate the situation (such as disabling TFO). Failing to proceed, the application (e.g., SSL library) may simply timeout and retry with TFO again, and the process repeats indefinitely. The proposed solution is to disable active TFO globally under the following circumstances: 1. client side TFO socket detects out of order FIN 2. client side TFO socket receives out of order RST We disable active side TFO globally for 1hr at first. Then if it happens again, we disable it for 2h, then 4h, 8h, ... And we reset the timeout to 1hr if a client side TFO sockets not opened on loopback has successfully received data segs from server. And we examine this condition during close(). The rational behind it is that when such firewall issue happens, application running on the client should eventually close the socket as it is not able to get the data it is expecting. Or application running on the server should close the socket as it is not able to receive any response from client. In both cases, out of order FIN or RST will get received on the client given that the firewall will not block them as no data are in those frames. And we want to disable active TFO globally as it helps if the middle box is very close to the client and most of the connections are likely to fail. Also, add a debug sysctl: tcp_fastopen_blackhole_detect_timeout_sec: the initial timeout to use when firewall blackhole issue happens. This can be set and read. When setting it to 0, it means to disable the active disable logic. Signed-off-by: Wei Wang <weiwan@google.com> Acked-by: Yuchung Cheng <ycheng@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-04-21 05:45:46 +08:00
bool tcp_fastopen_active_should_disable(struct sock *sk);
void tcp_fastopen_active_disable_ofo_check(struct sock *sk);
void tcp_fastopen_active_detect_blackhole(struct sock *sk, bool expired);
net/tcp_fastopen: Disable active side TFO in certain scenarios Middlebox firewall issues can potentially cause server's data being blackholed after a successful 3WHS using TFO. Following are the related reports from Apple: https://www.nanog.org/sites/default/files/Paasch_Network_Support.pdf Slide 31 identifies an issue where the client ACK to the server's data sent during a TFO'd handshake is dropped. C ---> syn-data ---> S C <--- syn/ack ----- S C (accept & write) C <---- data ------- S C ----- ACK -> X S [retry and timeout] https://www.ietf.org/proceedings/94/slides/slides-94-tcpm-13.pdf Slide 5 shows a similar situation that the server's data gets dropped after 3WHS. C ---- syn-data ---> S C <--- syn/ack ----- S C ---- ack --------> S S (accept & write) C? X <- data ------ S [retry and timeout] This is the worst failure b/c the client can not detect such behavior to mitigate the situation (such as disabling TFO). Failing to proceed, the application (e.g., SSL library) may simply timeout and retry with TFO again, and the process repeats indefinitely. The proposed solution is to disable active TFO globally under the following circumstances: 1. client side TFO socket detects out of order FIN 2. client side TFO socket receives out of order RST We disable active side TFO globally for 1hr at first. Then if it happens again, we disable it for 2h, then 4h, 8h, ... And we reset the timeout to 1hr if a client side TFO sockets not opened on loopback has successfully received data segs from server. And we examine this condition during close(). The rational behind it is that when such firewall issue happens, application running on the client should eventually close the socket as it is not able to get the data it is expecting. Or application running on the server should close the socket as it is not able to receive any response from client. In both cases, out of order FIN or RST will get received on the client given that the firewall will not block them as no data are in those frames. And we want to disable active TFO globally as it helps if the middle box is very close to the client and most of the connections are likely to fail. Also, add a debug sysctl: tcp_fastopen_blackhole_detect_timeout_sec: the initial timeout to use when firewall blackhole issue happens. This can be set and read. When setting it to 0, it means to disable the active disable logic. Signed-off-by: Wei Wang <weiwan@google.com> Acked-by: Yuchung Cheng <ycheng@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-04-21 05:45:46 +08:00
/* Caller needs to wrap with rcu_read_(un)lock() */
static inline
struct tcp_fastopen_context *tcp_fastopen_get_ctx(const struct sock *sk)
{
struct tcp_fastopen_context *ctx;
ctx = rcu_dereference(inet_csk(sk)->icsk_accept_queue.fastopenq.ctx);
if (!ctx)
ctx = rcu_dereference(sock_net(sk)->ipv4.tcp_fastopen_ctx);
return ctx;
}
static inline
bool tcp_fastopen_cookie_match(const struct tcp_fastopen_cookie *foc,
const struct tcp_fastopen_cookie *orig)
{
if (orig->len == TCP_FASTOPEN_COOKIE_SIZE &&
orig->len == foc->len &&
!memcmp(orig->val, foc->val, foc->len))
return true;
return false;
}
static inline
int tcp_fastopen_context_len(const struct tcp_fastopen_context *ctx)
{
net: ipv4: move tcp_fastopen server side code to SipHash library Using a bare block cipher in non-crypto code is almost always a bad idea, not only for security reasons (and we've seen some examples of this in the kernel in the past), but also for performance reasons. In the TCP fastopen case, we call into the bare AES block cipher one or two times (depending on whether the connection is IPv4 or IPv6). On most systems, this results in a call chain such as crypto_cipher_encrypt_one(ctx, dst, src) crypto_cipher_crt(tfm)->cit_encrypt_one(crypto_cipher_tfm(tfm), ...); aesni_encrypt kernel_fpu_begin(); aesni_enc(ctx, dst, src); // asm routine kernel_fpu_end(); It is highly unlikely that the use of special AES instructions has a benefit in this case, especially since we are doing the above twice for IPv6 connections, instead of using a transform which can process the entire input in one go. We could switch to the cbcmac(aes) shash, which would at least get rid of the duplicated overhead in *some* cases (i.e., today, only arm64 has an accelerated implementation of cbcmac(aes), while x86 will end up using the generic cbcmac template wrapping the AES-NI cipher, which basically ends up doing exactly the above). However, in the given context, it makes more sense to use a light-weight MAC algorithm that is more suitable for the purpose at hand, such as SipHash. Since the output size of SipHash already matches our chosen value for TCP_FASTOPEN_COOKIE_SIZE, and given that it accepts arbitrary input sizes, this greatly simplifies the code as well. NOTE: Server farms backing a single server IP for load balancing purposes and sharing a single fastopen key will be adversely affected by this change unless all systems in the pool receive their kernel upgrades at the same time. Signed-off-by: Ard Biesheuvel <ard.biesheuvel@linaro.org> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2019-06-17 16:09:33 +08:00
return ctx->num;
}
/* Latencies incurred by various limits for a sender. They are
* chronograph-like stats that are mutually exclusive.
*/
enum tcp_chrono {
TCP_CHRONO_UNSPEC,
TCP_CHRONO_BUSY, /* Actively sending data (non-empty write queue) */
TCP_CHRONO_RWND_LIMITED, /* Stalled by insufficient receive window */
TCP_CHRONO_SNDBUF_LIMITED, /* Stalled by insufficient send buffer */
__TCP_CHRONO_MAX,
};
void tcp_chrono_start(struct sock *sk, const enum tcp_chrono type);
void tcp_chrono_stop(struct sock *sk, const enum tcp_chrono type);
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 03:59:58 +08:00
/* This helper is needed, because skb->tcp_tsorted_anchor uses
* the same memory storage than skb->destructor/_skb_refdst
*/
static inline void tcp_skb_tsorted_anchor_cleanup(struct sk_buff *skb)
{
skb->destructor = NULL;
skb->_skb_refdst = 0UL;
}
#define tcp_skb_tsorted_save(skb) { \
unsigned long _save = skb->_skb_refdst; \
skb->_skb_refdst = 0UL;
#define tcp_skb_tsorted_restore(skb) \
skb->_skb_refdst = _save; \
}
void tcp_write_queue_purge(struct sock *sk);
tcp: implement rb-tree based retransmit queue Using a linear list to store all skbs in write queue has been okay for quite a while : O(N) is not too bad when N < 500. Things get messy when N is the order of 100,000 : Modern TCP stacks want 10Gbit+ of throughput even with 200 ms RTT flows. 40 ns per cache line miss means a full scan can use 4 ms, blowing away CPU caches. SACK processing often can use various hints to avoid parsing whole retransmit queue. But with high packet losses and/or high reordering, hints no longer work. Sender has to process thousands of unfriendly SACK, accumulating a huge socket backlog, burning a cpu and massively dropping packets. Using an rb-tree for retransmit queue has been avoided for years because it added complexity and overhead, but now is the time to be more resistant and say no to quadratic behavior. 1) RTX queue is no longer part of the write queue : already sent skbs are stored in one rb-tree. 2) Since reaching the head of write queue no longer needs sk->sk_send_head, we added an union of sk_send_head and tcp_rtx_queue Tested: On receiver : netem on ingress : delay 150ms 200us loss 1 GRO disabled to force stress and SACK storms. for f in `seq 1 10` do ./netperf -H lpaa6 -l30 -- -K bbr -o THROUGHPUT|tail -1 done | awk '{print $0} {sum += $0} END {printf "%7u\n",sum}' Before patch : 323.87 351.48 339.59 338.62 306.72 204.07 304.93 291.88 202.47 176.88 2840 After patch: 1700.83 2207.98 2070.17 1544.26 2114.76 2124.89 1693.14 1080.91 2216.82 1299.94 18053 Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-10-06 13:21:27 +08:00
static inline struct sk_buff *tcp_rtx_queue_head(const struct sock *sk)
{
return skb_rb_first(&sk->tcp_rtx_queue);
}
static inline struct sk_buff *tcp_rtx_queue_tail(const struct sock *sk)
{
return skb_rb_last(&sk->tcp_rtx_queue);
}
static inline struct sk_buff *tcp_write_queue_head(const struct sock *sk)
{
return skb_peek(&sk->sk_write_queue);
}
static inline struct sk_buff *tcp_write_queue_tail(const struct sock *sk)
{
return skb_peek_tail(&sk->sk_write_queue);
}
#define tcp_for_write_queue_from_safe(skb, tmp, sk) \
skb_queue_walk_from_safe(&(sk)->sk_write_queue, skb, tmp)
static inline struct sk_buff *tcp_send_head(const struct sock *sk)
{
tcp: implement rb-tree based retransmit queue Using a linear list to store all skbs in write queue has been okay for quite a while : O(N) is not too bad when N < 500. Things get messy when N is the order of 100,000 : Modern TCP stacks want 10Gbit+ of throughput even with 200 ms RTT flows. 40 ns per cache line miss means a full scan can use 4 ms, blowing away CPU caches. SACK processing often can use various hints to avoid parsing whole retransmit queue. But with high packet losses and/or high reordering, hints no longer work. Sender has to process thousands of unfriendly SACK, accumulating a huge socket backlog, burning a cpu and massively dropping packets. Using an rb-tree for retransmit queue has been avoided for years because it added complexity and overhead, but now is the time to be more resistant and say no to quadratic behavior. 1) RTX queue is no longer part of the write queue : already sent skbs are stored in one rb-tree. 2) Since reaching the head of write queue no longer needs sk->sk_send_head, we added an union of sk_send_head and tcp_rtx_queue Tested: On receiver : netem on ingress : delay 150ms 200us loss 1 GRO disabled to force stress and SACK storms. for f in `seq 1 10` do ./netperf -H lpaa6 -l30 -- -K bbr -o THROUGHPUT|tail -1 done | awk '{print $0} {sum += $0} END {printf "%7u\n",sum}' Before patch : 323.87 351.48 339.59 338.62 306.72 204.07 304.93 291.88 202.47 176.88 2840 After patch: 1700.83 2207.98 2070.17 1544.26 2114.76 2124.89 1693.14 1080.91 2216.82 1299.94 18053 Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-10-06 13:21:27 +08:00
return skb_peek(&sk->sk_write_queue);
}
static inline bool tcp_skb_is_last(const struct sock *sk,
const struct sk_buff *skb)
{
return skb_queue_is_last(&sk->sk_write_queue, skb);
}
/**
* tcp_write_queue_empty - test if any payload (or FIN) is available in write queue
* @sk: socket
*
* Since the write queue can have a temporary empty skb in it,
* we must not use "return skb_queue_empty(&sk->sk_write_queue)"
*/
tcp: implement rb-tree based retransmit queue Using a linear list to store all skbs in write queue has been okay for quite a while : O(N) is not too bad when N < 500. Things get messy when N is the order of 100,000 : Modern TCP stacks want 10Gbit+ of throughput even with 200 ms RTT flows. 40 ns per cache line miss means a full scan can use 4 ms, blowing away CPU caches. SACK processing often can use various hints to avoid parsing whole retransmit queue. But with high packet losses and/or high reordering, hints no longer work. Sender has to process thousands of unfriendly SACK, accumulating a huge socket backlog, burning a cpu and massively dropping packets. Using an rb-tree for retransmit queue has been avoided for years because it added complexity and overhead, but now is the time to be more resistant and say no to quadratic behavior. 1) RTX queue is no longer part of the write queue : already sent skbs are stored in one rb-tree. 2) Since reaching the head of write queue no longer needs sk->sk_send_head, we added an union of sk_send_head and tcp_rtx_queue Tested: On receiver : netem on ingress : delay 150ms 200us loss 1 GRO disabled to force stress and SACK storms. for f in `seq 1 10` do ./netperf -H lpaa6 -l30 -- -K bbr -o THROUGHPUT|tail -1 done | awk '{print $0} {sum += $0} END {printf "%7u\n",sum}' Before patch : 323.87 351.48 339.59 338.62 306.72 204.07 304.93 291.88 202.47 176.88 2840 After patch: 1700.83 2207.98 2070.17 1544.26 2114.76 2124.89 1693.14 1080.91 2216.82 1299.94 18053 Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-10-06 13:21:27 +08:00
static inline bool tcp_write_queue_empty(const struct sock *sk)
{
const struct tcp_sock *tp = tcp_sk(sk);
return tp->write_seq == tp->snd_nxt;
tcp: implement rb-tree based retransmit queue Using a linear list to store all skbs in write queue has been okay for quite a while : O(N) is not too bad when N < 500. Things get messy when N is the order of 100,000 : Modern TCP stacks want 10Gbit+ of throughput even with 200 ms RTT flows. 40 ns per cache line miss means a full scan can use 4 ms, blowing away CPU caches. SACK processing often can use various hints to avoid parsing whole retransmit queue. But with high packet losses and/or high reordering, hints no longer work. Sender has to process thousands of unfriendly SACK, accumulating a huge socket backlog, burning a cpu and massively dropping packets. Using an rb-tree for retransmit queue has been avoided for years because it added complexity and overhead, but now is the time to be more resistant and say no to quadratic behavior. 1) RTX queue is no longer part of the write queue : already sent skbs are stored in one rb-tree. 2) Since reaching the head of write queue no longer needs sk->sk_send_head, we added an union of sk_send_head and tcp_rtx_queue Tested: On receiver : netem on ingress : delay 150ms 200us loss 1 GRO disabled to force stress and SACK storms. for f in `seq 1 10` do ./netperf -H lpaa6 -l30 -- -K bbr -o THROUGHPUT|tail -1 done | awk '{print $0} {sum += $0} END {printf "%7u\n",sum}' Before patch : 323.87 351.48 339.59 338.62 306.72 204.07 304.93 291.88 202.47 176.88 2840 After patch: 1700.83 2207.98 2070.17 1544.26 2114.76 2124.89 1693.14 1080.91 2216.82 1299.94 18053 Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-10-06 13:21:27 +08:00
}
static inline bool tcp_rtx_queue_empty(const struct sock *sk)
{
return RB_EMPTY_ROOT(&sk->tcp_rtx_queue);
}
static inline bool tcp_rtx_and_write_queues_empty(const struct sock *sk)
{
return tcp_rtx_queue_empty(sk) && tcp_write_queue_empty(sk);
}
static inline void tcp_add_write_queue_tail(struct sock *sk, struct sk_buff *skb)
{
__skb_queue_tail(&sk->sk_write_queue, skb);
/* Queue it, remembering where we must start sending. */
tcp: highest_sack fix syzbot easily found a regression added in our latest patches [1] No longer set tp->highest_sack to the head of the send queue since this is not logical and error prone. Only sack processing should maintain the pointer to an skb from rtx queue. We might in the future only remember the sequence instead of a pointer to skb, since rb-tree should allow a fast lookup. [1] BUG: KASAN: use-after-free in tcp_highest_sack_seq include/net/tcp.h:1706 [inline] BUG: KASAN: use-after-free in tcp_ack+0x42bb/0x4fd0 net/ipv4/tcp_input.c:3537 Read of size 4 at addr ffff8801c154faa8 by task syz-executor4/12860 CPU: 0 PID: 12860 Comm: syz-executor4 Not tainted 4.14.0-next-20171113+ #41 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 print_address_description+0x73/0x250 mm/kasan/report.c:252 kasan_report_error mm/kasan/report.c:351 [inline] kasan_report+0x25b/0x340 mm/kasan/report.c:409 __asan_report_load4_noabort+0x14/0x20 mm/kasan/report.c:429 tcp_highest_sack_seq include/net/tcp.h:1706 [inline] tcp_ack+0x42bb/0x4fd0 net/ipv4/tcp_input.c:3537 tcp_rcv_established+0x672/0x18a0 net/ipv4/tcp_input.c:5439 tcp_v4_do_rcv+0x2ab/0x7d0 net/ipv4/tcp_ipv4.c:1468 sk_backlog_rcv include/net/sock.h:909 [inline] __release_sock+0x124/0x360 net/core/sock.c:2264 release_sock+0xa4/0x2a0 net/core/sock.c:2778 tcp_sendmsg+0x3a/0x50 net/ipv4/tcp.c:1462 inet_sendmsg+0x11f/0x5e0 net/ipv4/af_inet.c:763 sock_sendmsg_nosec net/socket.c:632 [inline] sock_sendmsg+0xca/0x110 net/socket.c:642 ___sys_sendmsg+0x75b/0x8a0 net/socket.c:2048 __sys_sendmsg+0xe5/0x210 net/socket.c:2082 SYSC_sendmsg net/socket.c:2093 [inline] SyS_sendmsg+0x2d/0x50 net/socket.c:2089 entry_SYSCALL_64_fastpath+0x1f/0x96 RIP: 0033:0x452879 RSP: 002b:00007fc9761bfbe8 EFLAGS: 00000212 ORIG_RAX: 000000000000002e RAX: ffffffffffffffda RBX: 0000000000758020 RCX: 0000000000452879 RDX: 0000000000000000 RSI: 0000000020917fc8 RDI: 0000000000000015 RBP: 0000000000000086 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000212 R12: 00000000006ee3a0 R13: 00000000ffffffff R14: 00007fc9761c06d4 R15: 0000000000000000 Allocated by task 12860: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:551 kasan_slab_alloc+0x12/0x20 mm/kasan/kasan.c:489 kmem_cache_alloc_node+0x144/0x760 mm/slab.c:3638 __alloc_skb+0xf1/0x780 net/core/skbuff.c:193 alloc_skb_fclone include/linux/skbuff.h:1023 [inline] sk_stream_alloc_skb+0x11d/0x900 net/ipv4/tcp.c:870 tcp_sendmsg_locked+0x1341/0x3b80 net/ipv4/tcp.c:1299 tcp_sendmsg+0x2f/0x50 net/ipv4/tcp.c:1461 inet_sendmsg+0x11f/0x5e0 net/ipv4/af_inet.c:763 sock_sendmsg_nosec net/socket.c:632 [inline] sock_sendmsg+0xca/0x110 net/socket.c:642 SYSC_sendto+0x358/0x5a0 net/socket.c:1749 SyS_sendto+0x40/0x50 net/socket.c:1717 entry_SYSCALL_64_fastpath+0x1f/0x96 Freed by task 12860: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_slab_free+0x71/0xc0 mm/kasan/kasan.c:524 __cache_free mm/slab.c:3492 [inline] kmem_cache_free+0x77/0x280 mm/slab.c:3750 kfree_skbmem+0xdd/0x1d0 net/core/skbuff.c:603 __kfree_skb+0x1d/0x20 net/core/skbuff.c:642 sk_wmem_free_skb include/net/sock.h:1419 [inline] tcp_rtx_queue_unlink_and_free include/net/tcp.h:1682 [inline] tcp_clean_rtx_queue net/ipv4/tcp_input.c:3111 [inline] tcp_ack+0x1b17/0x4fd0 net/ipv4/tcp_input.c:3593 tcp_rcv_established+0x672/0x18a0 net/ipv4/tcp_input.c:5439 tcp_v4_do_rcv+0x2ab/0x7d0 net/ipv4/tcp_ipv4.c:1468 sk_backlog_rcv include/net/sock.h:909 [inline] __release_sock+0x124/0x360 net/core/sock.c:2264 release_sock+0xa4/0x2a0 net/core/sock.c:2778 tcp_sendmsg+0x3a/0x50 net/ipv4/tcp.c:1462 inet_sendmsg+0x11f/0x5e0 net/ipv4/af_inet.c:763 sock_sendmsg_nosec net/socket.c:632 [inline] sock_sendmsg+0xca/0x110 net/socket.c:642 ___sys_sendmsg+0x75b/0x8a0 net/socket.c:2048 __sys_sendmsg+0xe5/0x210 net/socket.c:2082 SYSC_sendmsg net/socket.c:2093 [inline] SyS_sendmsg+0x2d/0x50 net/socket.c:2089 entry_SYSCALL_64_fastpath+0x1f/0x96 The buggy address belongs to the object at ffff8801c154fa80 which belongs to the cache skbuff_fclone_cache of size 456 The buggy address is located 40 bytes inside of 456-byte region [ffff8801c154fa80, ffff8801c154fc48) The buggy address belongs to the page: page:ffffea00070553c0 count:1 mapcount:0 mapping:ffff8801c154f080 index:0x0 flags: 0x2fffc0000000100(slab) raw: 02fffc0000000100 ffff8801c154f080 0000000000000000 0000000100000006 raw: ffffea00070a5a20 ffffea0006a18360 ffff8801d9ca0500 0000000000000000 page dumped because: kasan: bad access detected Fixes: 737ff314563c ("tcp: use sequence distance to detect reordering") Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Yuchung Cheng <ycheng@google.com> Reported-by: syzbot <syzkaller@googlegroups.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-11-15 13:02:19 +08:00
if (sk->sk_write_queue.next == skb)
tcp_chrono_start(sk, TCP_CHRONO_BUSY);
}
/* Insert new before skb on the write queue of sk. */
static inline void tcp_insert_write_queue_before(struct sk_buff *new,
struct sk_buff *skb,
struct sock *sk)
{
__skb_queue_before(&sk->sk_write_queue, skb, new);
}
static inline void tcp_unlink_write_queue(struct sk_buff *skb, struct sock *sk)
{
tcp_skb_tsorted_anchor_cleanup(skb);
__skb_unlink(skb, &sk->sk_write_queue);
}
tcp: implement rb-tree based retransmit queue Using a linear list to store all skbs in write queue has been okay for quite a while : O(N) is not too bad when N < 500. Things get messy when N is the order of 100,000 : Modern TCP stacks want 10Gbit+ of throughput even with 200 ms RTT flows. 40 ns per cache line miss means a full scan can use 4 ms, blowing away CPU caches. SACK processing often can use various hints to avoid parsing whole retransmit queue. But with high packet losses and/or high reordering, hints no longer work. Sender has to process thousands of unfriendly SACK, accumulating a huge socket backlog, burning a cpu and massively dropping packets. Using an rb-tree for retransmit queue has been avoided for years because it added complexity and overhead, but now is the time to be more resistant and say no to quadratic behavior. 1) RTX queue is no longer part of the write queue : already sent skbs are stored in one rb-tree. 2) Since reaching the head of write queue no longer needs sk->sk_send_head, we added an union of sk_send_head and tcp_rtx_queue Tested: On receiver : netem on ingress : delay 150ms 200us loss 1 GRO disabled to force stress and SACK storms. for f in `seq 1 10` do ./netperf -H lpaa6 -l30 -- -K bbr -o THROUGHPUT|tail -1 done | awk '{print $0} {sum += $0} END {printf "%7u\n",sum}' Before patch : 323.87 351.48 339.59 338.62 306.72 204.07 304.93 291.88 202.47 176.88 2840 After patch: 1700.83 2207.98 2070.17 1544.26 2114.76 2124.89 1693.14 1080.91 2216.82 1299.94 18053 Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-10-06 13:21:27 +08:00
void tcp_rbtree_insert(struct rb_root *root, struct sk_buff *skb);
static inline void tcp_rtx_queue_unlink(struct sk_buff *skb, struct sock *sk)
{
tcp: implement rb-tree based retransmit queue Using a linear list to store all skbs in write queue has been okay for quite a while : O(N) is not too bad when N < 500. Things get messy when N is the order of 100,000 : Modern TCP stacks want 10Gbit+ of throughput even with 200 ms RTT flows. 40 ns per cache line miss means a full scan can use 4 ms, blowing away CPU caches. SACK processing often can use various hints to avoid parsing whole retransmit queue. But with high packet losses and/or high reordering, hints no longer work. Sender has to process thousands of unfriendly SACK, accumulating a huge socket backlog, burning a cpu and massively dropping packets. Using an rb-tree for retransmit queue has been avoided for years because it added complexity and overhead, but now is the time to be more resistant and say no to quadratic behavior. 1) RTX queue is no longer part of the write queue : already sent skbs are stored in one rb-tree. 2) Since reaching the head of write queue no longer needs sk->sk_send_head, we added an union of sk_send_head and tcp_rtx_queue Tested: On receiver : netem on ingress : delay 150ms 200us loss 1 GRO disabled to force stress and SACK storms. for f in `seq 1 10` do ./netperf -H lpaa6 -l30 -- -K bbr -o THROUGHPUT|tail -1 done | awk '{print $0} {sum += $0} END {printf "%7u\n",sum}' Before patch : 323.87 351.48 339.59 338.62 306.72 204.07 304.93 291.88 202.47 176.88 2840 After patch: 1700.83 2207.98 2070.17 1544.26 2114.76 2124.89 1693.14 1080.91 2216.82 1299.94 18053 Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-10-06 13:21:27 +08:00
tcp_skb_tsorted_anchor_cleanup(skb);
rb_erase(&skb->rbnode, &sk->tcp_rtx_queue);
}
static inline void tcp_rtx_queue_unlink_and_free(struct sk_buff *skb, struct sock *sk)
{
list_del(&skb->tcp_tsorted_anchor);
tcp_rtx_queue_unlink(skb, sk);
sk_wmem_free_skb(sk, skb);
}
static inline void tcp_push_pending_frames(struct sock *sk)
{
if (tcp_send_head(sk)) {
struct tcp_sock *tp = tcp_sk(sk);
__tcp_push_pending_frames(sk, tcp_current_mss(sk), tp->nonagle);
}
}
/* Start sequence of the skb just after the highest skb with SACKed
* bit, valid only if sacked_out > 0 or when the caller has ensured
* validity by itself.
*/
static inline u32 tcp_highest_sack_seq(struct tcp_sock *tp)
{
if (!tp->sacked_out)
return tp->snd_una;
if (tp->highest_sack == NULL)
return tp->snd_nxt;
return TCP_SKB_CB(tp->highest_sack)->seq;
}
static inline void tcp_advance_highest_sack(struct sock *sk, struct sk_buff *skb)
{
tcp: highest_sack fix syzbot easily found a regression added in our latest patches [1] No longer set tp->highest_sack to the head of the send queue since this is not logical and error prone. Only sack processing should maintain the pointer to an skb from rtx queue. We might in the future only remember the sequence instead of a pointer to skb, since rb-tree should allow a fast lookup. [1] BUG: KASAN: use-after-free in tcp_highest_sack_seq include/net/tcp.h:1706 [inline] BUG: KASAN: use-after-free in tcp_ack+0x42bb/0x4fd0 net/ipv4/tcp_input.c:3537 Read of size 4 at addr ffff8801c154faa8 by task syz-executor4/12860 CPU: 0 PID: 12860 Comm: syz-executor4 Not tainted 4.14.0-next-20171113+ #41 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 print_address_description+0x73/0x250 mm/kasan/report.c:252 kasan_report_error mm/kasan/report.c:351 [inline] kasan_report+0x25b/0x340 mm/kasan/report.c:409 __asan_report_load4_noabort+0x14/0x20 mm/kasan/report.c:429 tcp_highest_sack_seq include/net/tcp.h:1706 [inline] tcp_ack+0x42bb/0x4fd0 net/ipv4/tcp_input.c:3537 tcp_rcv_established+0x672/0x18a0 net/ipv4/tcp_input.c:5439 tcp_v4_do_rcv+0x2ab/0x7d0 net/ipv4/tcp_ipv4.c:1468 sk_backlog_rcv include/net/sock.h:909 [inline] __release_sock+0x124/0x360 net/core/sock.c:2264 release_sock+0xa4/0x2a0 net/core/sock.c:2778 tcp_sendmsg+0x3a/0x50 net/ipv4/tcp.c:1462 inet_sendmsg+0x11f/0x5e0 net/ipv4/af_inet.c:763 sock_sendmsg_nosec net/socket.c:632 [inline] sock_sendmsg+0xca/0x110 net/socket.c:642 ___sys_sendmsg+0x75b/0x8a0 net/socket.c:2048 __sys_sendmsg+0xe5/0x210 net/socket.c:2082 SYSC_sendmsg net/socket.c:2093 [inline] SyS_sendmsg+0x2d/0x50 net/socket.c:2089 entry_SYSCALL_64_fastpath+0x1f/0x96 RIP: 0033:0x452879 RSP: 002b:00007fc9761bfbe8 EFLAGS: 00000212 ORIG_RAX: 000000000000002e RAX: ffffffffffffffda RBX: 0000000000758020 RCX: 0000000000452879 RDX: 0000000000000000 RSI: 0000000020917fc8 RDI: 0000000000000015 RBP: 0000000000000086 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000212 R12: 00000000006ee3a0 R13: 00000000ffffffff R14: 00007fc9761c06d4 R15: 0000000000000000 Allocated by task 12860: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:551 kasan_slab_alloc+0x12/0x20 mm/kasan/kasan.c:489 kmem_cache_alloc_node+0x144/0x760 mm/slab.c:3638 __alloc_skb+0xf1/0x780 net/core/skbuff.c:193 alloc_skb_fclone include/linux/skbuff.h:1023 [inline] sk_stream_alloc_skb+0x11d/0x900 net/ipv4/tcp.c:870 tcp_sendmsg_locked+0x1341/0x3b80 net/ipv4/tcp.c:1299 tcp_sendmsg+0x2f/0x50 net/ipv4/tcp.c:1461 inet_sendmsg+0x11f/0x5e0 net/ipv4/af_inet.c:763 sock_sendmsg_nosec net/socket.c:632 [inline] sock_sendmsg+0xca/0x110 net/socket.c:642 SYSC_sendto+0x358/0x5a0 net/socket.c:1749 SyS_sendto+0x40/0x50 net/socket.c:1717 entry_SYSCALL_64_fastpath+0x1f/0x96 Freed by task 12860: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_slab_free+0x71/0xc0 mm/kasan/kasan.c:524 __cache_free mm/slab.c:3492 [inline] kmem_cache_free+0x77/0x280 mm/slab.c:3750 kfree_skbmem+0xdd/0x1d0 net/core/skbuff.c:603 __kfree_skb+0x1d/0x20 net/core/skbuff.c:642 sk_wmem_free_skb include/net/sock.h:1419 [inline] tcp_rtx_queue_unlink_and_free include/net/tcp.h:1682 [inline] tcp_clean_rtx_queue net/ipv4/tcp_input.c:3111 [inline] tcp_ack+0x1b17/0x4fd0 net/ipv4/tcp_input.c:3593 tcp_rcv_established+0x672/0x18a0 net/ipv4/tcp_input.c:5439 tcp_v4_do_rcv+0x2ab/0x7d0 net/ipv4/tcp_ipv4.c:1468 sk_backlog_rcv include/net/sock.h:909 [inline] __release_sock+0x124/0x360 net/core/sock.c:2264 release_sock+0xa4/0x2a0 net/core/sock.c:2778 tcp_sendmsg+0x3a/0x50 net/ipv4/tcp.c:1462 inet_sendmsg+0x11f/0x5e0 net/ipv4/af_inet.c:763 sock_sendmsg_nosec net/socket.c:632 [inline] sock_sendmsg+0xca/0x110 net/socket.c:642 ___sys_sendmsg+0x75b/0x8a0 net/socket.c:2048 __sys_sendmsg+0xe5/0x210 net/socket.c:2082 SYSC_sendmsg net/socket.c:2093 [inline] SyS_sendmsg+0x2d/0x50 net/socket.c:2089 entry_SYSCALL_64_fastpath+0x1f/0x96 The buggy address belongs to the object at ffff8801c154fa80 which belongs to the cache skbuff_fclone_cache of size 456 The buggy address is located 40 bytes inside of 456-byte region [ffff8801c154fa80, ffff8801c154fc48) The buggy address belongs to the page: page:ffffea00070553c0 count:1 mapcount:0 mapping:ffff8801c154f080 index:0x0 flags: 0x2fffc0000000100(slab) raw: 02fffc0000000100 ffff8801c154f080 0000000000000000 0000000100000006 raw: ffffea00070a5a20 ffffea0006a18360 ffff8801d9ca0500 0000000000000000 page dumped because: kasan: bad access detected Fixes: 737ff314563c ("tcp: use sequence distance to detect reordering") Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Yuchung Cheng <ycheng@google.com> Reported-by: syzbot <syzkaller@googlegroups.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-11-15 13:02:19 +08:00
tcp_sk(sk)->highest_sack = skb_rb_next(skb);
}
static inline struct sk_buff *tcp_highest_sack(struct sock *sk)
{
return tcp_sk(sk)->highest_sack;
}
static inline void tcp_highest_sack_reset(struct sock *sk)
{
tcp: highest_sack fix syzbot easily found a regression added in our latest patches [1] No longer set tp->highest_sack to the head of the send queue since this is not logical and error prone. Only sack processing should maintain the pointer to an skb from rtx queue. We might in the future only remember the sequence instead of a pointer to skb, since rb-tree should allow a fast lookup. [1] BUG: KASAN: use-after-free in tcp_highest_sack_seq include/net/tcp.h:1706 [inline] BUG: KASAN: use-after-free in tcp_ack+0x42bb/0x4fd0 net/ipv4/tcp_input.c:3537 Read of size 4 at addr ffff8801c154faa8 by task syz-executor4/12860 CPU: 0 PID: 12860 Comm: syz-executor4 Not tainted 4.14.0-next-20171113+ #41 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 print_address_description+0x73/0x250 mm/kasan/report.c:252 kasan_report_error mm/kasan/report.c:351 [inline] kasan_report+0x25b/0x340 mm/kasan/report.c:409 __asan_report_load4_noabort+0x14/0x20 mm/kasan/report.c:429 tcp_highest_sack_seq include/net/tcp.h:1706 [inline] tcp_ack+0x42bb/0x4fd0 net/ipv4/tcp_input.c:3537 tcp_rcv_established+0x672/0x18a0 net/ipv4/tcp_input.c:5439 tcp_v4_do_rcv+0x2ab/0x7d0 net/ipv4/tcp_ipv4.c:1468 sk_backlog_rcv include/net/sock.h:909 [inline] __release_sock+0x124/0x360 net/core/sock.c:2264 release_sock+0xa4/0x2a0 net/core/sock.c:2778 tcp_sendmsg+0x3a/0x50 net/ipv4/tcp.c:1462 inet_sendmsg+0x11f/0x5e0 net/ipv4/af_inet.c:763 sock_sendmsg_nosec net/socket.c:632 [inline] sock_sendmsg+0xca/0x110 net/socket.c:642 ___sys_sendmsg+0x75b/0x8a0 net/socket.c:2048 __sys_sendmsg+0xe5/0x210 net/socket.c:2082 SYSC_sendmsg net/socket.c:2093 [inline] SyS_sendmsg+0x2d/0x50 net/socket.c:2089 entry_SYSCALL_64_fastpath+0x1f/0x96 RIP: 0033:0x452879 RSP: 002b:00007fc9761bfbe8 EFLAGS: 00000212 ORIG_RAX: 000000000000002e RAX: ffffffffffffffda RBX: 0000000000758020 RCX: 0000000000452879 RDX: 0000000000000000 RSI: 0000000020917fc8 RDI: 0000000000000015 RBP: 0000000000000086 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000212 R12: 00000000006ee3a0 R13: 00000000ffffffff R14: 00007fc9761c06d4 R15: 0000000000000000 Allocated by task 12860: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:551 kasan_slab_alloc+0x12/0x20 mm/kasan/kasan.c:489 kmem_cache_alloc_node+0x144/0x760 mm/slab.c:3638 __alloc_skb+0xf1/0x780 net/core/skbuff.c:193 alloc_skb_fclone include/linux/skbuff.h:1023 [inline] sk_stream_alloc_skb+0x11d/0x900 net/ipv4/tcp.c:870 tcp_sendmsg_locked+0x1341/0x3b80 net/ipv4/tcp.c:1299 tcp_sendmsg+0x2f/0x50 net/ipv4/tcp.c:1461 inet_sendmsg+0x11f/0x5e0 net/ipv4/af_inet.c:763 sock_sendmsg_nosec net/socket.c:632 [inline] sock_sendmsg+0xca/0x110 net/socket.c:642 SYSC_sendto+0x358/0x5a0 net/socket.c:1749 SyS_sendto+0x40/0x50 net/socket.c:1717 entry_SYSCALL_64_fastpath+0x1f/0x96 Freed by task 12860: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_slab_free+0x71/0xc0 mm/kasan/kasan.c:524 __cache_free mm/slab.c:3492 [inline] kmem_cache_free+0x77/0x280 mm/slab.c:3750 kfree_skbmem+0xdd/0x1d0 net/core/skbuff.c:603 __kfree_skb+0x1d/0x20 net/core/skbuff.c:642 sk_wmem_free_skb include/net/sock.h:1419 [inline] tcp_rtx_queue_unlink_and_free include/net/tcp.h:1682 [inline] tcp_clean_rtx_queue net/ipv4/tcp_input.c:3111 [inline] tcp_ack+0x1b17/0x4fd0 net/ipv4/tcp_input.c:3593 tcp_rcv_established+0x672/0x18a0 net/ipv4/tcp_input.c:5439 tcp_v4_do_rcv+0x2ab/0x7d0 net/ipv4/tcp_ipv4.c:1468 sk_backlog_rcv include/net/sock.h:909 [inline] __release_sock+0x124/0x360 net/core/sock.c:2264 release_sock+0xa4/0x2a0 net/core/sock.c:2778 tcp_sendmsg+0x3a/0x50 net/ipv4/tcp.c:1462 inet_sendmsg+0x11f/0x5e0 net/ipv4/af_inet.c:763 sock_sendmsg_nosec net/socket.c:632 [inline] sock_sendmsg+0xca/0x110 net/socket.c:642 ___sys_sendmsg+0x75b/0x8a0 net/socket.c:2048 __sys_sendmsg+0xe5/0x210 net/socket.c:2082 SYSC_sendmsg net/socket.c:2093 [inline] SyS_sendmsg+0x2d/0x50 net/socket.c:2089 entry_SYSCALL_64_fastpath+0x1f/0x96 The buggy address belongs to the object at ffff8801c154fa80 which belongs to the cache skbuff_fclone_cache of size 456 The buggy address is located 40 bytes inside of 456-byte region [ffff8801c154fa80, ffff8801c154fc48) The buggy address belongs to the page: page:ffffea00070553c0 count:1 mapcount:0 mapping:ffff8801c154f080 index:0x0 flags: 0x2fffc0000000100(slab) raw: 02fffc0000000100 ffff8801c154f080 0000000000000000 0000000100000006 raw: ffffea00070a5a20 ffffea0006a18360 ffff8801d9ca0500 0000000000000000 page dumped because: kasan: bad access detected Fixes: 737ff314563c ("tcp: use sequence distance to detect reordering") Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Yuchung Cheng <ycheng@google.com> Reported-by: syzbot <syzkaller@googlegroups.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-11-15 13:02:19 +08:00
tcp_sk(sk)->highest_sack = tcp_rtx_queue_head(sk);
}
/* Called when old skb is about to be deleted and replaced by new skb */
static inline void tcp_highest_sack_replace(struct sock *sk,
struct sk_buff *old,
struct sk_buff *new)
{
if (old == tcp_highest_sack(sk))
tcp_sk(sk)->highest_sack = new;
}
/* This helper checks if socket has IP_TRANSPARENT set */
static inline bool inet_sk_transparent(const struct sock *sk)
{
switch (sk->sk_state) {
case TCP_TIME_WAIT:
return inet_twsk(sk)->tw_transparent;
case TCP_NEW_SYN_RECV:
return inet_rsk(inet_reqsk(sk))->no_srccheck;
}
return inet_sk(sk)->transparent;
}
/* Determines whether this is a thin stream (which may suffer from
* increased latency). Used to trigger latency-reducing mechanisms.
*/
static inline bool tcp_stream_is_thin(struct tcp_sock *tp)
{
return tp->packets_out < 4 && !tcp_in_initial_slowstart(tp);
}
/* /proc */
enum tcp_seq_states {
TCP_SEQ_STATE_LISTENING,
TCP_SEQ_STATE_ESTABLISHED,
};
void *tcp_seq_start(struct seq_file *seq, loff_t *pos);
void *tcp_seq_next(struct seq_file *seq, void *v, loff_t *pos);
void tcp_seq_stop(struct seq_file *seq, void *v);
struct tcp_seq_afinfo {
sa_family_t family;
};
struct tcp_iter_state {
struct seq_net_private p;
enum tcp_seq_states state;
struct sock *syn_wait_sk;
struct tcp_seq_afinfo *bpf_seq_afinfo;
int bucket, offset, sbucket, num;
loff_t last_pos;
};
extern struct request_sock_ops tcp_request_sock_ops;
extern struct request_sock_ops tcp6_request_sock_ops;
void tcp_v4_destroy_sock(struct sock *sk);
struct sk_buff *tcp_gso_segment(struct sk_buff *skb,
netdev_features_t features);
struct sk_buff *tcp_gro_receive(struct list_head *head, struct sk_buff *skb);
INDIRECT_CALLABLE_DECLARE(int tcp4_gro_complete(struct sk_buff *skb, int thoff));
INDIRECT_CALLABLE_DECLARE(struct sk_buff *tcp4_gro_receive(struct list_head *head, struct sk_buff *skb));
INDIRECT_CALLABLE_DECLARE(int tcp6_gro_complete(struct sk_buff *skb, int thoff));
INDIRECT_CALLABLE_DECLARE(struct sk_buff *tcp6_gro_receive(struct list_head *head, struct sk_buff *skb));
int tcp_gro_complete(struct sk_buff *skb);
void __tcp_v4_send_check(struct sk_buff *skb, __be32 saddr, __be32 daddr);
tcp: TCP_NOTSENT_LOWAT socket option Idea of this patch is to add optional limitation of number of unsent bytes in TCP sockets, to reduce usage of kernel memory. TCP receiver might announce a big window, and TCP sender autotuning might allow a large amount of bytes in write queue, but this has little performance impact if a large part of this buffering is wasted : Write queue needs to be large only to deal with large BDP, not necessarily to cope with scheduling delays (incoming ACKS make room for the application to queue more bytes) For most workloads, using a value of 128 KB or less is OK to give applications enough time to react to POLLOUT events in time (or being awaken in a blocking sendmsg()) This patch adds two ways to set the limit : 1) Per socket option TCP_NOTSENT_LOWAT 2) A sysctl (/proc/sys/net/ipv4/tcp_notsent_lowat) for sockets not using TCP_NOTSENT_LOWAT socket option (or setting a zero value) Default value being UINT_MAX (0xFFFFFFFF), meaning this has no effect. This changes poll()/select()/epoll() to report POLLOUT only if number of unsent bytes is below tp->nosent_lowat Note this might increase number of sendmsg()/sendfile() calls when using non blocking sockets, and increase number of context switches for blocking sockets. Note this is not related to SO_SNDLOWAT (as SO_SNDLOWAT is defined as : Specify the minimum number of bytes in the buffer until the socket layer will pass the data to the protocol) Tested: netperf sessions, and watching /proc/net/protocols "memory" column for TCP With 200 concurrent netperf -t TCP_STREAM sessions, amount of kernel memory used by TCP buffers shrinks by ~55 % (20567 pages instead of 45458) lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols TCPv6 1880 2 45458 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y TCP 1696 508 45458 no 208 yes kernel y y y y y y y y y y y y y n y y y y y lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols TCPv6 1880 2 20567 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y TCP 1696 508 20567 no 208 yes kernel y y y y y y y y y y y y y n y y y y y Using 128KB has no bad effect on the throughput or cpu usage of a single flow, although there is an increase of context switches. A bonus is that we hold socket lock for a shorter amount of time and should improve latencies of ACK processing. lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3 OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf. Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand Size Size Size (sec) Util Util Util Util Demand Demand Units Final Final % Method % Method 1651584 6291456 16384 20.00 17447.90 10^6bits/s 3.13 S -1.00 U 0.353 -1.000 usec/KB Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3': 412,514 context-switches 200.034645535 seconds time elapsed lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3 OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf. Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand Size Size Size (sec) Util Util Util Util Demand Demand Units Final Final % Method % Method 1593240 6291456 16384 20.00 17321.16 10^6bits/s 3.35 S -1.00 U 0.381 -1.000 usec/KB Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3': 2,675,818 context-switches 200.029651391 seconds time elapsed Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-By: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2013-07-23 11:27:07 +08:00
static inline u32 tcp_notsent_lowat(const struct tcp_sock *tp)
{
struct net *net = sock_net((struct sock *)tp);
return tp->notsent_lowat ?: net->ipv4.sysctl_tcp_notsent_lowat;
tcp: TCP_NOTSENT_LOWAT socket option Idea of this patch is to add optional limitation of number of unsent bytes in TCP sockets, to reduce usage of kernel memory. TCP receiver might announce a big window, and TCP sender autotuning might allow a large amount of bytes in write queue, but this has little performance impact if a large part of this buffering is wasted : Write queue needs to be large only to deal with large BDP, not necessarily to cope with scheduling delays (incoming ACKS make room for the application to queue more bytes) For most workloads, using a value of 128 KB or less is OK to give applications enough time to react to POLLOUT events in time (or being awaken in a blocking sendmsg()) This patch adds two ways to set the limit : 1) Per socket option TCP_NOTSENT_LOWAT 2) A sysctl (/proc/sys/net/ipv4/tcp_notsent_lowat) for sockets not using TCP_NOTSENT_LOWAT socket option (or setting a zero value) Default value being UINT_MAX (0xFFFFFFFF), meaning this has no effect. This changes poll()/select()/epoll() to report POLLOUT only if number of unsent bytes is below tp->nosent_lowat Note this might increase number of sendmsg()/sendfile() calls when using non blocking sockets, and increase number of context switches for blocking sockets. Note this is not related to SO_SNDLOWAT (as SO_SNDLOWAT is defined as : Specify the minimum number of bytes in the buffer until the socket layer will pass the data to the protocol) Tested: netperf sessions, and watching /proc/net/protocols "memory" column for TCP With 200 concurrent netperf -t TCP_STREAM sessions, amount of kernel memory used by TCP buffers shrinks by ~55 % (20567 pages instead of 45458) lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols TCPv6 1880 2 45458 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y TCP 1696 508 45458 no 208 yes kernel y y y y y y y y y y y y y n y y y y y lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols TCPv6 1880 2 20567 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y TCP 1696 508 20567 no 208 yes kernel y y y y y y y y y y y y y n y y y y y Using 128KB has no bad effect on the throughput or cpu usage of a single flow, although there is an increase of context switches. A bonus is that we hold socket lock for a shorter amount of time and should improve latencies of ACK processing. lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3 OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf. Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand Size Size Size (sec) Util Util Util Util Demand Demand Units Final Final % Method % Method 1651584 6291456 16384 20.00 17447.90 10^6bits/s 3.13 S -1.00 U 0.353 -1.000 usec/KB Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3': 412,514 context-switches 200.034645535 seconds time elapsed lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3 OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf. Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand Size Size Size (sec) Util Util Util Util Demand Demand Units Final Final % Method % Method 1593240 6291456 16384 20.00 17321.16 10^6bits/s 3.35 S -1.00 U 0.381 -1.000 usec/KB Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3': 2,675,818 context-switches 200.029651391 seconds time elapsed Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-By: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2013-07-23 11:27:07 +08:00
}
bool tcp_stream_memory_free(const struct sock *sk, int wake);
tcp: TCP_NOTSENT_LOWAT socket option Idea of this patch is to add optional limitation of number of unsent bytes in TCP sockets, to reduce usage of kernel memory. TCP receiver might announce a big window, and TCP sender autotuning might allow a large amount of bytes in write queue, but this has little performance impact if a large part of this buffering is wasted : Write queue needs to be large only to deal with large BDP, not necessarily to cope with scheduling delays (incoming ACKS make room for the application to queue more bytes) For most workloads, using a value of 128 KB or less is OK to give applications enough time to react to POLLOUT events in time (or being awaken in a blocking sendmsg()) This patch adds two ways to set the limit : 1) Per socket option TCP_NOTSENT_LOWAT 2) A sysctl (/proc/sys/net/ipv4/tcp_notsent_lowat) for sockets not using TCP_NOTSENT_LOWAT socket option (or setting a zero value) Default value being UINT_MAX (0xFFFFFFFF), meaning this has no effect. This changes poll()/select()/epoll() to report POLLOUT only if number of unsent bytes is below tp->nosent_lowat Note this might increase number of sendmsg()/sendfile() calls when using non blocking sockets, and increase number of context switches for blocking sockets. Note this is not related to SO_SNDLOWAT (as SO_SNDLOWAT is defined as : Specify the minimum number of bytes in the buffer until the socket layer will pass the data to the protocol) Tested: netperf sessions, and watching /proc/net/protocols "memory" column for TCP With 200 concurrent netperf -t TCP_STREAM sessions, amount of kernel memory used by TCP buffers shrinks by ~55 % (20567 pages instead of 45458) lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols TCPv6 1880 2 45458 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y TCP 1696 508 45458 no 208 yes kernel y y y y y y y y y y y y y n y y y y y lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols TCPv6 1880 2 20567 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y TCP 1696 508 20567 no 208 yes kernel y y y y y y y y y y y y y n y y y y y Using 128KB has no bad effect on the throughput or cpu usage of a single flow, although there is an increase of context switches. A bonus is that we hold socket lock for a shorter amount of time and should improve latencies of ACK processing. lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3 OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf. Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand Size Size Size (sec) Util Util Util Util Demand Demand Units Final Final % Method % Method 1651584 6291456 16384 20.00 17447.90 10^6bits/s 3.13 S -1.00 U 0.353 -1.000 usec/KB Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3': 412,514 context-switches 200.034645535 seconds time elapsed lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3 OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf. Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand Size Size Size (sec) Util Util Util Util Demand Demand Units Final Final % Method % Method 1593240 6291456 16384 20.00 17321.16 10^6bits/s 3.35 S -1.00 U 0.381 -1.000 usec/KB Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3': 2,675,818 context-switches 200.029651391 seconds time elapsed Signed-off-by: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-By: Yuchung Cheng <ycheng@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2013-07-23 11:27:07 +08:00
#ifdef CONFIG_PROC_FS
int tcp4_proc_init(void);
void tcp4_proc_exit(void);
#endif
int tcp_rtx_synack(const struct sock *sk, struct request_sock *req);
int tcp_conn_request(struct request_sock_ops *rsk_ops,
const struct tcp_request_sock_ops *af_ops,
struct sock *sk, struct sk_buff *skb);
/* TCP af-specific functions */
struct tcp_sock_af_ops {
#ifdef CONFIG_TCP_MD5SIG
struct tcp_md5sig_key *(*md5_lookup) (const struct sock *sk,
const struct sock *addr_sk);
int (*calc_md5_hash)(char *location,
const struct tcp_md5sig_key *md5,
const struct sock *sk,
const struct sk_buff *skb);
int (*md5_parse)(struct sock *sk,
int optname,
sockptr_t optval,
int optlen);
#endif
};
struct tcp_request_sock_ops {
u16 mss_clamp;
#ifdef CONFIG_TCP_MD5SIG
struct tcp_md5sig_key *(*req_md5_lookup)(const struct sock *sk,
const struct sock *addr_sk);
int (*calc_md5_hash) (char *location,
const struct tcp_md5sig_key *md5,
const struct sock *sk,
const struct sk_buff *skb);
#endif
void (*init_req)(struct request_sock *req,
const struct sock *sk_listener,
struct sk_buff *skb);
#ifdef CONFIG_SYN_COOKIES
__u32 (*cookie_init_seq)(const struct sk_buff *skb,
__u16 *mss);
#endif
struct dst_entry *(*route_req)(const struct sock *sk, struct flowi *fl,
const struct request_sock *req);
u32 (*init_seq)(const struct sk_buff *skb);
u32 (*init_ts_off)(const struct net *net, const struct sk_buff *skb);
int (*send_synack)(const struct sock *sk, struct dst_entry *dst,
struct flowi *fl, struct request_sock *req,
struct tcp_fastopen_cookie *foc,
bpf: tcp: Add bpf_skops_hdr_opt_len() and bpf_skops_write_hdr_opt() The bpf prog needs to parse the SYN header to learn what options have been sent by the peer's bpf-prog before writing its options into SYNACK. This patch adds a "syn_skb" arg to tcp_make_synack() and send_synack(). This syn_skb will eventually be made available (as read-only) to the bpf prog. This will be the only SYN packet available to the bpf prog during syncookie. For other regular cases, the bpf prog can also use the saved_syn. When writing options, the bpf prog will first be called to tell the kernel its required number of bytes. It is done by the new bpf_skops_hdr_opt_len(). The bpf prog will only be called when the new BPF_SOCK_OPS_WRITE_HDR_OPT_CB_FLAG is set in tp->bpf_sock_ops_cb_flags. When the bpf prog returns, the kernel will know how many bytes are needed and then update the "*remaining" arg accordingly. 4 byte alignment will be included in the "*remaining" before this function returns. The 4 byte aligned number of bytes will also be stored into the opts->bpf_opt_len. "bpf_opt_len" is a newly added member to the struct tcp_out_options. Then the new bpf_skops_write_hdr_opt() will call the bpf prog to write the header options. The bpf prog is only called if it has reserved spaces before (opts->bpf_opt_len > 0). The bpf prog is the last one getting a chance to reserve header space and writing the header option. These two functions are half implemented to highlight the changes in TCP stack. The actual codes preparing the bpf running context and invoking the bpf prog will be added in the later patch with other necessary bpf pieces. Signed-off-by: Martin KaFai Lau <kafai@fb.com> Signed-off-by: Alexei Starovoitov <ast@kernel.org> Reviewed-by: Eric Dumazet <edumazet@google.com> Link: https://lore.kernel.org/bpf/20200820190052.2885316-1-kafai@fb.com
2020-08-21 03:00:52 +08:00
enum tcp_synack_type synack_type,
struct sk_buff *syn_skb);
};
extern const struct tcp_request_sock_ops tcp_request_sock_ipv4_ops;
#if IS_ENABLED(CONFIG_IPV6)
extern const struct tcp_request_sock_ops tcp_request_sock_ipv6_ops;
#endif
#ifdef CONFIG_SYN_COOKIES
static inline __u32 cookie_init_sequence(const struct tcp_request_sock_ops *ops,
const struct sock *sk, struct sk_buff *skb,
__u16 *mss)
{
tcp_synq_overflow(sk);
__NET_INC_STATS(sock_net(sk), LINUX_MIB_SYNCOOKIESSENT);
return ops->cookie_init_seq(skb, mss);
}
#else
static inline __u32 cookie_init_sequence(const struct tcp_request_sock_ops *ops,
const struct sock *sk, struct sk_buff *skb,
__u16 *mss)
{
return 0;
}
#endif
int tcpv4_offload_init(void);
void tcp_v4_init(void);
void tcp_init(void);
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 12:57:46 +08:00
/* tcp_recovery.c */
void tcp_mark_skb_lost(struct sock *sk, struct sk_buff *skb);
void tcp_newreno_mark_lost(struct sock *sk, bool snd_una_advanced);
extern s32 tcp_rack_skb_timeout(struct tcp_sock *tp, struct sk_buff *skb,
u32 reo_wnd);
extern void tcp_rack_mark_lost(struct sock *sk);
extern void tcp_rack_advance(struct tcp_sock *tp, u8 sacked, u32 end_seq,
u64 xmit_time);
tcp: add reordering timer in RACK loss detection This patch makes RACK install a reordering timer when it suspects some packets might be lost, but wants to delay the decision a little bit to accomodate reordering. It does not create a new timer but instead repurposes the existing RTO timer, because both are meant to retransmit packets. Specifically it arms a timer ICSK_TIME_REO_TIMEOUT when the RACK timing check fails. The wait time is set to RACK.RTT + RACK.reo_wnd - (NOW - Packet.xmit_time) + fudge This translates to expecting a packet (Packet) should take (RACK.RTT + RACK.reo_wnd + fudge) to deliver after it was sent. When there are multiple packets that need a timer, we use one timer with the maximum timeout. Therefore the timer conservatively uses the maximum window to expire N packets by one timeout, instead of N timeouts to expire N packets sent at different times. The fudge factor is 2 jiffies to ensure when the timer fires, all the suspected packets would exceed the deadline and be marked lost by tcp_rack_detect_loss(). It has to be at least 1 jiffy because the clock may tick between calling icsk_reset_xmit_timer(timeout) and actually hang the timer. The next jiffy is to lower-bound the timeout to 2 jiffies when reo_wnd is < 1ms. When the reordering timer fires (tcp_rack_reo_timeout): If we aren't in Recovery we'll enter fast recovery and force fast retransmit. This is very similar to the early retransmit (RFC5827) except RACK is not constrained to only enter recovery for small outstanding flights. Signed-off-by: Yuchung Cheng <ycheng@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-01-13 14:11:33 +08:00
extern void tcp_rack_reo_timeout(struct sock *sk);
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 07:38:48 +08:00
extern void tcp_rack_update_reo_wnd(struct sock *sk, struct rate_sample *rs);
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 12:57:46 +08:00
/* At how many usecs into the future should the RTO fire? */
static inline s64 tcp_rto_delta_us(const struct sock *sk)
{
tcp: implement rb-tree based retransmit queue Using a linear list to store all skbs in write queue has been okay for quite a while : O(N) is not too bad when N < 500. Things get messy when N is the order of 100,000 : Modern TCP stacks want 10Gbit+ of throughput even with 200 ms RTT flows. 40 ns per cache line miss means a full scan can use 4 ms, blowing away CPU caches. SACK processing often can use various hints to avoid parsing whole retransmit queue. But with high packet losses and/or high reordering, hints no longer work. Sender has to process thousands of unfriendly SACK, accumulating a huge socket backlog, burning a cpu and massively dropping packets. Using an rb-tree for retransmit queue has been avoided for years because it added complexity and overhead, but now is the time to be more resistant and say no to quadratic behavior. 1) RTX queue is no longer part of the write queue : already sent skbs are stored in one rb-tree. 2) Since reaching the head of write queue no longer needs sk->sk_send_head, we added an union of sk_send_head and tcp_rtx_queue Tested: On receiver : netem on ingress : delay 150ms 200us loss 1 GRO disabled to force stress and SACK storms. for f in `seq 1 10` do ./netperf -H lpaa6 -l30 -- -K bbr -o THROUGHPUT|tail -1 done | awk '{print $0} {sum += $0} END {printf "%7u\n",sum}' Before patch : 323.87 351.48 339.59 338.62 306.72 204.07 304.93 291.88 202.47 176.88 2840 After patch: 1700.83 2207.98 2070.17 1544.26 2114.76 2124.89 1693.14 1080.91 2216.82 1299.94 18053 Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-10-06 13:21:27 +08:00
const struct sk_buff *skb = tcp_rtx_queue_head(sk);
u32 rto = inet_csk(sk)->icsk_rto;
u64 rto_time_stamp_us = tcp_skb_timestamp_us(skb) + jiffies_to_usecs(rto);
return rto_time_stamp_us - tcp_sk(sk)->tcp_mstamp;
}
/*
* Save and compile IPv4 options, return a pointer to it
*/
static inline struct ip_options_rcu *tcp_v4_save_options(struct net *net,
struct sk_buff *skb)
{
const struct ip_options *opt = &TCP_SKB_CB(skb)->header.h4.opt;
struct ip_options_rcu *dopt = NULL;
if (opt->optlen) {
int opt_size = sizeof(*dopt) + opt->optlen;
dopt = kmalloc(opt_size, GFP_ATOMIC);
if (dopt && __ip_options_echo(net, &dopt->opt, skb, opt)) {
kfree(dopt);
dopt = NULL;
}
}
return dopt;
}
/* locally generated TCP pure ACKs have skb->truesize == 2
* (check tcp_send_ack() in net/ipv4/tcp_output.c )
* This is much faster than dissecting the packet to find out.
* (Think of GRE encapsulations, IPv4, IPv6, ...)
*/
static inline bool skb_is_tcp_pure_ack(const struct sk_buff *skb)
{
return skb->truesize == 2;
}
static inline void skb_set_tcp_pure_ack(struct sk_buff *skb)
{
skb->truesize = 2;
}
static inline int tcp_inq(struct sock *sk)
{
struct tcp_sock *tp = tcp_sk(sk);
int answ;
if ((1 << sk->sk_state) & (TCPF_SYN_SENT | TCPF_SYN_RECV)) {
answ = 0;
} else if (sock_flag(sk, SOCK_URGINLINE) ||
!tp->urg_data ||
before(tp->urg_seq, tp->copied_seq) ||
!before(tp->urg_seq, tp->rcv_nxt)) {
answ = tp->rcv_nxt - tp->copied_seq;
/* Subtract 1, if FIN was received */
if (answ && sock_flag(sk, SOCK_DONE))
answ--;
} else {
answ = tp->urg_seq - tp->copied_seq;
}
return answ;
}
int tcp_peek_len(struct socket *sock);
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 01:52:15 +08:00
static inline void tcp_segs_in(struct tcp_sock *tp, const struct sk_buff *skb)
{
u16 segs_in;
segs_in = max_t(u16, 1, skb_shinfo(skb)->gso_segs);
tp->segs_in += segs_in;
if (skb->len > tcp_hdrlen(skb))
tp->data_segs_in += segs_in;
}
/*
* TCP listen path runs lockless.
* We forced "struct sock" to be const qualified to make sure
* we don't modify one of its field by mistake.
* Here, we increment sk_drops which is an atomic_t, so we can safely
* make sock writable again.
*/
static inline void tcp_listendrop(const struct sock *sk)
{
atomic_inc(&((struct sock *)sk)->sk_drops);
__NET_INC_STATS(sock_net(sk), LINUX_MIB_LISTENDROPS);
}
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 19:24:36 +08:00
enum hrtimer_restart tcp_pace_kick(struct hrtimer *timer);
/*
* Interface for adding Upper Level Protocols over TCP
*/
#define TCP_ULP_NAME_MAX 16
#define TCP_ULP_MAX 128
#define TCP_ULP_BUF_MAX (TCP_ULP_NAME_MAX*TCP_ULP_MAX)
struct tcp_ulp_ops {
struct list_head list;
/* initialize ulp */
int (*init)(struct sock *sk);
bpf: sockmap/tls, close can race with map free When a map free is called and in parallel a socket is closed we have two paths that can potentially reset the socket prot ops, the bpf close() path and the map free path. This creates a problem with which prot ops should be used from the socket closed side. If the map_free side completes first then we want to call the original lowest level ops. However, if the tls path runs first we want to call the sockmap ops. Additionally there was no locking around prot updates in TLS code paths so the prot ops could be changed multiple times once from TLS path and again from sockmap side potentially leaving ops pointed at either TLS or sockmap when psock and/or tls context have already been destroyed. To fix this race first only update ops inside callback lock so that TLS, sockmap and lowest level all agree on prot state. Second and a ULP callback update() so that lower layers can inform the upper layer when they are being removed allowing the upper layer to reset prot ops. This gets us close to allowing sockmap and tls to be stacked in arbitrary order but will save that patch for *next trees. v4: - make sure we don't free things for device; - remove the checks which swap the callbacks back only if TLS is at the top. Reported-by: syzbot+06537213db7ba2745c4a@syzkaller.appspotmail.com Fixes: 02c558b2d5d6 ("bpf: sockmap, support for msg_peek in sk_msg with redirect ingress") Signed-off-by: John Fastabend <john.fastabend@gmail.com> Signed-off-by: Jakub Kicinski <jakub.kicinski@netronome.com> Reviewed-by: Dirk van der Merwe <dirk.vandermerwe@netronome.com> Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
2019-07-20 01:29:22 +08:00
/* update ulp */
void (*update)(struct sock *sk, struct proto *p,
void (*write_space)(struct sock *sk));
/* cleanup ulp */
void (*release)(struct sock *sk);
/* diagnostic */
int (*get_info)(const struct sock *sk, struct sk_buff *skb);
size_t (*get_info_size)(const struct sock *sk);
/* clone ulp */
void (*clone)(const struct request_sock *req, struct sock *newsk,
const gfp_t priority);
char name[TCP_ULP_NAME_MAX];
struct module *owner;
};
int tcp_register_ulp(struct tcp_ulp_ops *type);
void tcp_unregister_ulp(struct tcp_ulp_ops *type);
int tcp_set_ulp(struct sock *sk, const char *name);
void tcp_get_available_ulp(char *buf, size_t len);
void tcp_cleanup_ulp(struct sock *sk);
void tcp_update_ulp(struct sock *sk, struct proto *p,
void (*write_space)(struct sock *sk));
#define MODULE_ALIAS_TCP_ULP(name) \
__MODULE_INFO(alias, alias_userspace, name); \
__MODULE_INFO(alias, alias_tcp_ulp, "tcp-ulp-" name)
bpf, sockmap: convert to generic sk_msg interface Add a generic sk_msg layer, and convert current sockmap and later kTLS over to make use of it. While sk_buff handles network packet representation from netdevice up to socket, sk_msg handles data representation from application to socket layer. This means that sk_msg framework spans across ULP users in the kernel, and enables features such as introspection or filtering of data with the help of BPF programs that operate on this data structure. Latter becomes in particular useful for kTLS where data encryption is deferred into the kernel, and as such enabling the kernel to perform L7 introspection and policy based on BPF for TLS connections where the record is being encrypted after BPF has run and came to a verdict. In order to get there, first step is to transform open coding of scatter-gather list handling into a common core framework that subsystems can use. The code itself has been split and refactored into three bigger pieces: i) the generic sk_msg API which deals with managing the scatter gather ring, providing helpers for walking and mangling, transferring application data from user space into it, and preparing it for BPF pre/post-processing, ii) the plain sock map itself where sockets can be attached to or detached from; these bits are independent of i) which can now be used also without sock map, and iii) the integration with plain TCP as one protocol to be used for processing L7 application data (later this could e.g. also be extended to other protocols like UDP). The semantics are the same with the old sock map code and therefore no change of user facing behavior or APIs. While pursuing this work it also helped finding a number of bugs in the old sockmap code that we've fixed already in earlier commits. The test_sockmap kselftest suite passes through fine as well. Joint work with John. Signed-off-by: Daniel Borkmann <daniel@iogearbox.net> Signed-off-by: John Fastabend <john.fastabend@gmail.com> Signed-off-by: Alexei Starovoitov <ast@kernel.org>
2018-10-13 08:45:58 +08:00
struct sk_msg;
struct sk_psock;
#ifdef CONFIG_BPF_STREAM_PARSER
struct proto *tcp_bpf_get_proto(struct sock *sk, struct sk_psock *psock);
void tcp_bpf_clone(const struct sock *sk, struct sock *newsk);
#else
static inline void tcp_bpf_clone(const struct sock *sk, struct sock *newsk)
{
}
#endif /* CONFIG_BPF_STREAM_PARSER */
#ifdef CONFIG_NET_SOCK_MSG
bpf, sockmap: convert to generic sk_msg interface Add a generic sk_msg layer, and convert current sockmap and later kTLS over to make use of it. While sk_buff handles network packet representation from netdevice up to socket, sk_msg handles data representation from application to socket layer. This means that sk_msg framework spans across ULP users in the kernel, and enables features such as introspection or filtering of data with the help of BPF programs that operate on this data structure. Latter becomes in particular useful for kTLS where data encryption is deferred into the kernel, and as such enabling the kernel to perform L7 introspection and policy based on BPF for TLS connections where the record is being encrypted after BPF has run and came to a verdict. In order to get there, first step is to transform open coding of scatter-gather list handling into a common core framework that subsystems can use. The code itself has been split and refactored into three bigger pieces: i) the generic sk_msg API which deals with managing the scatter gather ring, providing helpers for walking and mangling, transferring application data from user space into it, and preparing it for BPF pre/post-processing, ii) the plain sock map itself where sockets can be attached to or detached from; these bits are independent of i) which can now be used also without sock map, and iii) the integration with plain TCP as one protocol to be used for processing L7 application data (later this could e.g. also be extended to other protocols like UDP). The semantics are the same with the old sock map code and therefore no change of user facing behavior or APIs. While pursuing this work it also helped finding a number of bugs in the old sockmap code that we've fixed already in earlier commits. The test_sockmap kselftest suite passes through fine as well. Joint work with John. Signed-off-by: Daniel Borkmann <daniel@iogearbox.net> Signed-off-by: John Fastabend <john.fastabend@gmail.com> Signed-off-by: Alexei Starovoitov <ast@kernel.org>
2018-10-13 08:45:58 +08:00
int tcp_bpf_sendmsg_redir(struct sock *sk, struct sk_msg *msg, u32 bytes,
int flags);
int __tcp_bpf_recvmsg(struct sock *sk, struct sk_psock *psock,
struct msghdr *msg, int len, int flags);
#endif /* CONFIG_NET_SOCK_MSG */
bpf, sockmap: convert to generic sk_msg interface Add a generic sk_msg layer, and convert current sockmap and later kTLS over to make use of it. While sk_buff handles network packet representation from netdevice up to socket, sk_msg handles data representation from application to socket layer. This means that sk_msg framework spans across ULP users in the kernel, and enables features such as introspection or filtering of data with the help of BPF programs that operate on this data structure. Latter becomes in particular useful for kTLS where data encryption is deferred into the kernel, and as such enabling the kernel to perform L7 introspection and policy based on BPF for TLS connections where the record is being encrypted after BPF has run and came to a verdict. In order to get there, first step is to transform open coding of scatter-gather list handling into a common core framework that subsystems can use. The code itself has been split and refactored into three bigger pieces: i) the generic sk_msg API which deals with managing the scatter gather ring, providing helpers for walking and mangling, transferring application data from user space into it, and preparing it for BPF pre/post-processing, ii) the plain sock map itself where sockets can be attached to or detached from; these bits are independent of i) which can now be used also without sock map, and iii) the integration with plain TCP as one protocol to be used for processing L7 application data (later this could e.g. also be extended to other protocols like UDP). The semantics are the same with the old sock map code and therefore no change of user facing behavior or APIs. While pursuing this work it also helped finding a number of bugs in the old sockmap code that we've fixed already in earlier commits. The test_sockmap kselftest suite passes through fine as well. Joint work with John. Signed-off-by: Daniel Borkmann <daniel@iogearbox.net> Signed-off-by: John Fastabend <john.fastabend@gmail.com> Signed-off-by: Alexei Starovoitov <ast@kernel.org>
2018-10-13 08:45:58 +08:00
bpf: tcp: Allow bpf prog to write and parse TCP header option [ Note: The TCP changes here is mainly to implement the bpf pieces into the bpf_skops_*() functions introduced in the earlier patches. ] The earlier effort in BPF-TCP-CC allows the TCP Congestion Control algorithm to be written in BPF. It opens up opportunities to allow a faster turnaround time in testing/releasing new congestion control ideas to production environment. The same flexibility can be extended to writing TCP header option. It is not uncommon that people want to test new TCP header option to improve the TCP performance. Another use case is for data-center that has a more controlled environment and has more flexibility in putting header options for internal only use. For example, we want to test the idea in putting maximum delay ACK in TCP header option which is similar to a draft RFC proposal [1]. This patch introduces the necessary BPF API and use them in the TCP stack to allow BPF_PROG_TYPE_SOCK_OPS program to parse and write TCP header options. It currently supports most of the TCP packet except RST. Supported TCP header option: ─────────────────────────── This patch allows the bpf-prog to write any option kind. Different bpf-progs can write its own option by calling the new helper bpf_store_hdr_opt(). The helper will ensure there is no duplicated option in the header. By allowing bpf-prog to write any option kind, this gives a lot of flexibility to the bpf-prog. Different bpf-prog can write its own option kind. It could also allow the bpf-prog to support a recently standardized option on an older kernel. Sockops Callback Flags: ────────────────────── The bpf program will only be called to parse/write tcp header option if the following newly added callback flags are enabled in tp->bpf_sock_ops_cb_flags: BPF_SOCK_OPS_PARSE_UNKNOWN_HDR_OPT_CB_FLAG BPF_SOCK_OPS_PARSE_ALL_HDR_OPT_CB_FLAG BPF_SOCK_OPS_WRITE_HDR_OPT_CB_FLAG A few words on the PARSE CB flags. When the above PARSE CB flags are turned on, the bpf-prog will be called on packets received at a sk that has at least reached the ESTABLISHED state. The parsing of the SYN-SYNACK-ACK will be discussed in the "3 Way HandShake" section. The default is off for all of the above new CB flags, i.e. the bpf prog will not be called to parse or write bpf hdr option. There are details comment on these new cb flags in the UAPI bpf.h. sock_ops->skb_data and bpf_load_hdr_opt() ───────────────────────────────────────── sock_ops->skb_data and sock_ops->skb_data_end covers the whole TCP header and its options. They are read only. The new bpf_load_hdr_opt() helps to read a particular option "kind" from the skb_data. Please refer to the comment in UAPI bpf.h. It has details on what skb_data contains under different sock_ops->op. 3 Way HandShake ─────────────── The bpf-prog can learn if it is sending SYN or SYNACK by reading the sock_ops->skb_tcp_flags. * Passive side When writing SYNACK (i.e. sock_ops->op == BPF_SOCK_OPS_WRITE_HDR_OPT_CB), the received SYN skb will be available to the bpf prog. The bpf prog can use the SYN skb (which may carry the header option sent from the remote bpf prog) to decide what bpf header option should be written to the outgoing SYNACK skb. The SYN packet can be obtained by getsockopt(TCP_BPF_SYN*). More on this later. Also, the bpf prog can learn if it is in syncookie mode (by checking sock_ops->args[0] == BPF_WRITE_HDR_TCP_SYNACK_COOKIE). The bpf prog can store the received SYN pkt by using the existing bpf_setsockopt(TCP_SAVE_SYN). The example in a later patch does it. [ Note that the fullsock here is a listen sk, bpf_sk_storage is not very useful here since the listen sk will be shared by many concurrent connection requests. Extending bpf_sk_storage support to request_sock will add weight to the minisock and it is not necessary better than storing the whole ~100 bytes SYN pkt. ] When the connection is established, the bpf prog will be called in the existing PASSIVE_ESTABLISHED_CB callback. At that time, the bpf prog can get the header option from the saved syn and then apply the needed operation to the newly established socket. The later patch will use the max delay ack specified in the SYN header and set the RTO of this newly established connection as an example. The received ACK (that concludes the 3WHS) will also be available to the bpf prog during PASSIVE_ESTABLISHED_CB through the sock_ops->skb_data. It could be useful in syncookie scenario. More on this later. There is an existing getsockopt "TCP_SAVED_SYN" to return the whole saved syn pkt which includes the IP[46] header and the TCP header. A few "TCP_BPF_SYN*" getsockopt has been added to allow specifying where to start getting from, e.g. starting from TCP header, or from IP[46] header. The new getsockopt(TCP_BPF_SYN*) will also know where it can get the SYN's packet from: - (a) the just received syn (available when the bpf prog is writing SYNACK) and it is the only way to get SYN during syncookie mode. or - (b) the saved syn (available in PASSIVE_ESTABLISHED_CB and also other existing CB). The bpf prog does not need to know where the SYN pkt is coming from. The getsockopt(TCP_BPF_SYN*) will hide this details. Similarly, a flags "BPF_LOAD_HDR_OPT_TCP_SYN" is also added to bpf_load_hdr_opt() to read a particular header option from the SYN packet. * Fastopen Fastopen should work the same as the regular non fastopen case. This is a test in a later patch. * Syncookie For syncookie, the later example patch asks the active side's bpf prog to resend the header options in ACK. The server can use bpf_load_hdr_opt() to look at the options in this received ACK during PASSIVE_ESTABLISHED_CB. * Active side The bpf prog will get a chance to write the bpf header option in the SYN packet during WRITE_HDR_OPT_CB. The received SYNACK pkt will also be available to the bpf prog during the existing ACTIVE_ESTABLISHED_CB callback through the sock_ops->skb_data and bpf_load_hdr_opt(). * Turn off header CB flags after 3WHS If the bpf prog does not need to write/parse header options beyond the 3WHS, the bpf prog can clear the bpf_sock_ops_cb_flags to avoid being called for header options. Or the bpf-prog can select to leave the UNKNOWN_HDR_OPT_CB_FLAG on so that the kernel will only call it when there is option that the kernel cannot handle. [1]: draft-wang-tcpm-low-latency-opt-00 https://tools.ietf.org/html/draft-wang-tcpm-low-latency-opt-00 Signed-off-by: Martin KaFai Lau <kafai@fb.com> Signed-off-by: Alexei Starovoitov <ast@kernel.org> Link: https://lore.kernel.org/bpf/20200820190104.2885895-1-kafai@fb.com
2020-08-21 03:01:04 +08:00
#ifdef CONFIG_CGROUP_BPF
static inline void bpf_skops_init_skb(struct bpf_sock_ops_kern *skops,
struct sk_buff *skb,
unsigned int end_offset)
{
skops->skb = skb;
skops->skb_data_end = skb->data + end_offset;
}
#else
static inline void bpf_skops_init_skb(struct bpf_sock_ops_kern *skops,
struct sk_buff *skb,
unsigned int end_offset)
{
}
#endif
bpf: BPF support for sock_ops Created a new BPF program type, BPF_PROG_TYPE_SOCK_OPS, and a corresponding struct that allows BPF programs of this type to access some of the socket's fields (such as IP addresses, ports, etc.). It uses the existing bpf cgroups infrastructure so the programs can be attached per cgroup with full inheritance support. The program will be called at appropriate times to set relevant connections parameters such as buffer sizes, SYN and SYN-ACK RTOs, etc., based on connection information such as IP addresses, port numbers, etc. Alghough there are already 3 mechanisms to set parameters (sysctls, route metrics and setsockopts), this new mechanism provides some distinct advantages. Unlike sysctls, it can set parameters per connection. In contrast to route metrics, it can also use port numbers and information provided by a user level program. In addition, it could set parameters probabilistically for evaluation purposes (i.e. do something different on 10% of the flows and compare results with the other 90% of the flows). Also, in cases where IPv6 addresses contain geographic information, the rules to make changes based on the distance (or RTT) between the hosts are much easier than route metric rules and can be global. Finally, unlike setsockopt, it oes not require application changes and it can be updated easily at any time. Although the bpf cgroup framework already contains a sock related program type (BPF_PROG_TYPE_CGROUP_SOCK), I created the new type (BPF_PROG_TYPE_SOCK_OPS) beccause the existing type expects to be called only once during the connections's lifetime. In contrast, the new program type will be called multiple times from different places in the network stack code. For example, before sending SYN and SYN-ACKs to set an appropriate timeout, when the connection is established to set congestion control, etc. As a result it has "op" field to specify the type of operation requested. The purpose of this new program type is to simplify setting connection parameters, such as buffer sizes, TCP's SYN RTO, etc. For example, it is easy to use facebook's internal IPv6 addresses to determine if both hosts of a connection are in the same datacenter. Therefore, it is easy to write a BPF program to choose a small SYN RTO value when both hosts are in the same datacenter. This patch only contains the framework to support the new BPF program type, following patches add the functionality to set various connection parameters. This patch defines a new BPF program type: BPF_PROG_TYPE_SOCKET_OPS and a new bpf syscall command to load a new program of this type: BPF_PROG_LOAD_SOCKET_OPS. Two new corresponding structs (one for the kernel one for the user/BPF program): /* kernel version */ struct bpf_sock_ops_kern { struct sock *sk; __u32 op; union { __u32 reply; __u32 replylong[4]; }; }; /* user version * Some fields are in network byte order reflecting the sock struct * Use the bpf_ntohl helper macro in samples/bpf/bpf_endian.h to * convert them to host byte order. */ struct bpf_sock_ops { __u32 op; union { __u32 reply; __u32 replylong[4]; }; __u32 family; __u32 remote_ip4; /* In network byte order */ __u32 local_ip4; /* In network byte order */ __u32 remote_ip6[4]; /* In network byte order */ __u32 local_ip6[4]; /* In network byte order */ __u32 remote_port; /* In network byte order */ __u32 local_port; /* In host byte horder */ }; Currently there are two types of ops. The first type expects the BPF program to return a value which is then used by the caller (or a negative value to indicate the operation is not supported). The second type expects state changes to be done by the BPF program, for example through a setsockopt BPF helper function, and they ignore the return value. The reply fields of the bpf_sockt_ops struct are there in case a bpf program needs to return a value larger than an integer. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Daniel Borkmann <daniel@iogearbox.net> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-07-01 11:02:40 +08:00
/* Call BPF_SOCK_OPS program that returns an int. If the return value
* is < 0, then the BPF op failed (for example if the loaded BPF
* program does not support the chosen operation or there is no BPF
* program loaded).
*/
#ifdef CONFIG_BPF
static inline int tcp_call_bpf(struct sock *sk, int op, u32 nargs, u32 *args)
bpf: BPF support for sock_ops Created a new BPF program type, BPF_PROG_TYPE_SOCK_OPS, and a corresponding struct that allows BPF programs of this type to access some of the socket's fields (such as IP addresses, ports, etc.). It uses the existing bpf cgroups infrastructure so the programs can be attached per cgroup with full inheritance support. The program will be called at appropriate times to set relevant connections parameters such as buffer sizes, SYN and SYN-ACK RTOs, etc., based on connection information such as IP addresses, port numbers, etc. Alghough there are already 3 mechanisms to set parameters (sysctls, route metrics and setsockopts), this new mechanism provides some distinct advantages. Unlike sysctls, it can set parameters per connection. In contrast to route metrics, it can also use port numbers and information provided by a user level program. In addition, it could set parameters probabilistically for evaluation purposes (i.e. do something different on 10% of the flows and compare results with the other 90% of the flows). Also, in cases where IPv6 addresses contain geographic information, the rules to make changes based on the distance (or RTT) between the hosts are much easier than route metric rules and can be global. Finally, unlike setsockopt, it oes not require application changes and it can be updated easily at any time. Although the bpf cgroup framework already contains a sock related program type (BPF_PROG_TYPE_CGROUP_SOCK), I created the new type (BPF_PROG_TYPE_SOCK_OPS) beccause the existing type expects to be called only once during the connections's lifetime. In contrast, the new program type will be called multiple times from different places in the network stack code. For example, before sending SYN and SYN-ACKs to set an appropriate timeout, when the connection is established to set congestion control, etc. As a result it has "op" field to specify the type of operation requested. The purpose of this new program type is to simplify setting connection parameters, such as buffer sizes, TCP's SYN RTO, etc. For example, it is easy to use facebook's internal IPv6 addresses to determine if both hosts of a connection are in the same datacenter. Therefore, it is easy to write a BPF program to choose a small SYN RTO value when both hosts are in the same datacenter. This patch only contains the framework to support the new BPF program type, following patches add the functionality to set various connection parameters. This patch defines a new BPF program type: BPF_PROG_TYPE_SOCKET_OPS and a new bpf syscall command to load a new program of this type: BPF_PROG_LOAD_SOCKET_OPS. Two new corresponding structs (one for the kernel one for the user/BPF program): /* kernel version */ struct bpf_sock_ops_kern { struct sock *sk; __u32 op; union { __u32 reply; __u32 replylong[4]; }; }; /* user version * Some fields are in network byte order reflecting the sock struct * Use the bpf_ntohl helper macro in samples/bpf/bpf_endian.h to * convert them to host byte order. */ struct bpf_sock_ops { __u32 op; union { __u32 reply; __u32 replylong[4]; }; __u32 family; __u32 remote_ip4; /* In network byte order */ __u32 local_ip4; /* In network byte order */ __u32 remote_ip6[4]; /* In network byte order */ __u32 local_ip6[4]; /* In network byte order */ __u32 remote_port; /* In network byte order */ __u32 local_port; /* In host byte horder */ }; Currently there are two types of ops. The first type expects the BPF program to return a value which is then used by the caller (or a negative value to indicate the operation is not supported). The second type expects state changes to be done by the BPF program, for example through a setsockopt BPF helper function, and they ignore the return value. The reply fields of the bpf_sockt_ops struct are there in case a bpf program needs to return a value larger than an integer. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Daniel Borkmann <daniel@iogearbox.net> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-07-01 11:02:40 +08:00
{
struct bpf_sock_ops_kern sock_ops;
int ret;
memset(&sock_ops, 0, offsetof(struct bpf_sock_ops_kern, temp));
if (sk_fullsock(sk)) {
sock_ops.is_fullsock = 1;
bpf: BPF support for sock_ops Created a new BPF program type, BPF_PROG_TYPE_SOCK_OPS, and a corresponding struct that allows BPF programs of this type to access some of the socket's fields (such as IP addresses, ports, etc.). It uses the existing bpf cgroups infrastructure so the programs can be attached per cgroup with full inheritance support. The program will be called at appropriate times to set relevant connections parameters such as buffer sizes, SYN and SYN-ACK RTOs, etc., based on connection information such as IP addresses, port numbers, etc. Alghough there are already 3 mechanisms to set parameters (sysctls, route metrics and setsockopts), this new mechanism provides some distinct advantages. Unlike sysctls, it can set parameters per connection. In contrast to route metrics, it can also use port numbers and information provided by a user level program. In addition, it could set parameters probabilistically for evaluation purposes (i.e. do something different on 10% of the flows and compare results with the other 90% of the flows). Also, in cases where IPv6 addresses contain geographic information, the rules to make changes based on the distance (or RTT) between the hosts are much easier than route metric rules and can be global. Finally, unlike setsockopt, it oes not require application changes and it can be updated easily at any time. Although the bpf cgroup framework already contains a sock related program type (BPF_PROG_TYPE_CGROUP_SOCK), I created the new type (BPF_PROG_TYPE_SOCK_OPS) beccause the existing type expects to be called only once during the connections's lifetime. In contrast, the new program type will be called multiple times from different places in the network stack code. For example, before sending SYN and SYN-ACKs to set an appropriate timeout, when the connection is established to set congestion control, etc. As a result it has "op" field to specify the type of operation requested. The purpose of this new program type is to simplify setting connection parameters, such as buffer sizes, TCP's SYN RTO, etc. For example, it is easy to use facebook's internal IPv6 addresses to determine if both hosts of a connection are in the same datacenter. Therefore, it is easy to write a BPF program to choose a small SYN RTO value when both hosts are in the same datacenter. This patch only contains the framework to support the new BPF program type, following patches add the functionality to set various connection parameters. This patch defines a new BPF program type: BPF_PROG_TYPE_SOCKET_OPS and a new bpf syscall command to load a new program of this type: BPF_PROG_LOAD_SOCKET_OPS. Two new corresponding structs (one for the kernel one for the user/BPF program): /* kernel version */ struct bpf_sock_ops_kern { struct sock *sk; __u32 op; union { __u32 reply; __u32 replylong[4]; }; }; /* user version * Some fields are in network byte order reflecting the sock struct * Use the bpf_ntohl helper macro in samples/bpf/bpf_endian.h to * convert them to host byte order. */ struct bpf_sock_ops { __u32 op; union { __u32 reply; __u32 replylong[4]; }; __u32 family; __u32 remote_ip4; /* In network byte order */ __u32 local_ip4; /* In network byte order */ __u32 remote_ip6[4]; /* In network byte order */ __u32 local_ip6[4]; /* In network byte order */ __u32 remote_port; /* In network byte order */ __u32 local_port; /* In host byte horder */ }; Currently there are two types of ops. The first type expects the BPF program to return a value which is then used by the caller (or a negative value to indicate the operation is not supported). The second type expects state changes to be done by the BPF program, for example through a setsockopt BPF helper function, and they ignore the return value. The reply fields of the bpf_sockt_ops struct are there in case a bpf program needs to return a value larger than an integer. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Daniel Borkmann <daniel@iogearbox.net> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-07-01 11:02:40 +08:00
sock_owned_by_me(sk);
}
bpf: BPF support for sock_ops Created a new BPF program type, BPF_PROG_TYPE_SOCK_OPS, and a corresponding struct that allows BPF programs of this type to access some of the socket's fields (such as IP addresses, ports, etc.). It uses the existing bpf cgroups infrastructure so the programs can be attached per cgroup with full inheritance support. The program will be called at appropriate times to set relevant connections parameters such as buffer sizes, SYN and SYN-ACK RTOs, etc., based on connection information such as IP addresses, port numbers, etc. Alghough there are already 3 mechanisms to set parameters (sysctls, route metrics and setsockopts), this new mechanism provides some distinct advantages. Unlike sysctls, it can set parameters per connection. In contrast to route metrics, it can also use port numbers and information provided by a user level program. In addition, it could set parameters probabilistically for evaluation purposes (i.e. do something different on 10% of the flows and compare results with the other 90% of the flows). Also, in cases where IPv6 addresses contain geographic information, the rules to make changes based on the distance (or RTT) between the hosts are much easier than route metric rules and can be global. Finally, unlike setsockopt, it oes not require application changes and it can be updated easily at any time. Although the bpf cgroup framework already contains a sock related program type (BPF_PROG_TYPE_CGROUP_SOCK), I created the new type (BPF_PROG_TYPE_SOCK_OPS) beccause the existing type expects to be called only once during the connections's lifetime. In contrast, the new program type will be called multiple times from different places in the network stack code. For example, before sending SYN and SYN-ACKs to set an appropriate timeout, when the connection is established to set congestion control, etc. As a result it has "op" field to specify the type of operation requested. The purpose of this new program type is to simplify setting connection parameters, such as buffer sizes, TCP's SYN RTO, etc. For example, it is easy to use facebook's internal IPv6 addresses to determine if both hosts of a connection are in the same datacenter. Therefore, it is easy to write a BPF program to choose a small SYN RTO value when both hosts are in the same datacenter. This patch only contains the framework to support the new BPF program type, following patches add the functionality to set various connection parameters. This patch defines a new BPF program type: BPF_PROG_TYPE_SOCKET_OPS and a new bpf syscall command to load a new program of this type: BPF_PROG_LOAD_SOCKET_OPS. Two new corresponding structs (one for the kernel one for the user/BPF program): /* kernel version */ struct bpf_sock_ops_kern { struct sock *sk; __u32 op; union { __u32 reply; __u32 replylong[4]; }; }; /* user version * Some fields are in network byte order reflecting the sock struct * Use the bpf_ntohl helper macro in samples/bpf/bpf_endian.h to * convert them to host byte order. */ struct bpf_sock_ops { __u32 op; union { __u32 reply; __u32 replylong[4]; }; __u32 family; __u32 remote_ip4; /* In network byte order */ __u32 local_ip4; /* In network byte order */ __u32 remote_ip6[4]; /* In network byte order */ __u32 local_ip6[4]; /* In network byte order */ __u32 remote_port; /* In network byte order */ __u32 local_port; /* In host byte horder */ }; Currently there are two types of ops. The first type expects the BPF program to return a value which is then used by the caller (or a negative value to indicate the operation is not supported). The second type expects state changes to be done by the BPF program, for example through a setsockopt BPF helper function, and they ignore the return value. The reply fields of the bpf_sockt_ops struct are there in case a bpf program needs to return a value larger than an integer. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Daniel Borkmann <daniel@iogearbox.net> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-07-01 11:02:40 +08:00
sock_ops.sk = sk;
sock_ops.op = op;
if (nargs > 0)
memcpy(sock_ops.args, args, nargs * sizeof(*args));
bpf: BPF support for sock_ops Created a new BPF program type, BPF_PROG_TYPE_SOCK_OPS, and a corresponding struct that allows BPF programs of this type to access some of the socket's fields (such as IP addresses, ports, etc.). It uses the existing bpf cgroups infrastructure so the programs can be attached per cgroup with full inheritance support. The program will be called at appropriate times to set relevant connections parameters such as buffer sizes, SYN and SYN-ACK RTOs, etc., based on connection information such as IP addresses, port numbers, etc. Alghough there are already 3 mechanisms to set parameters (sysctls, route metrics and setsockopts), this new mechanism provides some distinct advantages. Unlike sysctls, it can set parameters per connection. In contrast to route metrics, it can also use port numbers and information provided by a user level program. In addition, it could set parameters probabilistically for evaluation purposes (i.e. do something different on 10% of the flows and compare results with the other 90% of the flows). Also, in cases where IPv6 addresses contain geographic information, the rules to make changes based on the distance (or RTT) between the hosts are much easier than route metric rules and can be global. Finally, unlike setsockopt, it oes not require application changes and it can be updated easily at any time. Although the bpf cgroup framework already contains a sock related program type (BPF_PROG_TYPE_CGROUP_SOCK), I created the new type (BPF_PROG_TYPE_SOCK_OPS) beccause the existing type expects to be called only once during the connections's lifetime. In contrast, the new program type will be called multiple times from different places in the network stack code. For example, before sending SYN and SYN-ACKs to set an appropriate timeout, when the connection is established to set congestion control, etc. As a result it has "op" field to specify the type of operation requested. The purpose of this new program type is to simplify setting connection parameters, such as buffer sizes, TCP's SYN RTO, etc. For example, it is easy to use facebook's internal IPv6 addresses to determine if both hosts of a connection are in the same datacenter. Therefore, it is easy to write a BPF program to choose a small SYN RTO value when both hosts are in the same datacenter. This patch only contains the framework to support the new BPF program type, following patches add the functionality to set various connection parameters. This patch defines a new BPF program type: BPF_PROG_TYPE_SOCKET_OPS and a new bpf syscall command to load a new program of this type: BPF_PROG_LOAD_SOCKET_OPS. Two new corresponding structs (one for the kernel one for the user/BPF program): /* kernel version */ struct bpf_sock_ops_kern { struct sock *sk; __u32 op; union { __u32 reply; __u32 replylong[4]; }; }; /* user version * Some fields are in network byte order reflecting the sock struct * Use the bpf_ntohl helper macro in samples/bpf/bpf_endian.h to * convert them to host byte order. */ struct bpf_sock_ops { __u32 op; union { __u32 reply; __u32 replylong[4]; }; __u32 family; __u32 remote_ip4; /* In network byte order */ __u32 local_ip4; /* In network byte order */ __u32 remote_ip6[4]; /* In network byte order */ __u32 local_ip6[4]; /* In network byte order */ __u32 remote_port; /* In network byte order */ __u32 local_port; /* In host byte horder */ }; Currently there are two types of ops. The first type expects the BPF program to return a value which is then used by the caller (or a negative value to indicate the operation is not supported). The second type expects state changes to be done by the BPF program, for example through a setsockopt BPF helper function, and they ignore the return value. The reply fields of the bpf_sockt_ops struct are there in case a bpf program needs to return a value larger than an integer. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Daniel Borkmann <daniel@iogearbox.net> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-07-01 11:02:40 +08:00
ret = BPF_CGROUP_RUN_PROG_SOCK_OPS(&sock_ops);
if (ret == 0)
ret = sock_ops.reply;
else
ret = -1;
return ret;
}
static inline int tcp_call_bpf_2arg(struct sock *sk, int op, u32 arg1, u32 arg2)
{
u32 args[2] = {arg1, arg2};
return tcp_call_bpf(sk, op, 2, args);
}
static inline int tcp_call_bpf_3arg(struct sock *sk, int op, u32 arg1, u32 arg2,
u32 arg3)
{
u32 args[3] = {arg1, arg2, arg3};
return tcp_call_bpf(sk, op, 3, args);
}
bpf: BPF support for sock_ops Created a new BPF program type, BPF_PROG_TYPE_SOCK_OPS, and a corresponding struct that allows BPF programs of this type to access some of the socket's fields (such as IP addresses, ports, etc.). It uses the existing bpf cgroups infrastructure so the programs can be attached per cgroup with full inheritance support. The program will be called at appropriate times to set relevant connections parameters such as buffer sizes, SYN and SYN-ACK RTOs, etc., based on connection information such as IP addresses, port numbers, etc. Alghough there are already 3 mechanisms to set parameters (sysctls, route metrics and setsockopts), this new mechanism provides some distinct advantages. Unlike sysctls, it can set parameters per connection. In contrast to route metrics, it can also use port numbers and information provided by a user level program. In addition, it could set parameters probabilistically for evaluation purposes (i.e. do something different on 10% of the flows and compare results with the other 90% of the flows). Also, in cases where IPv6 addresses contain geographic information, the rules to make changes based on the distance (or RTT) between the hosts are much easier than route metric rules and can be global. Finally, unlike setsockopt, it oes not require application changes and it can be updated easily at any time. Although the bpf cgroup framework already contains a sock related program type (BPF_PROG_TYPE_CGROUP_SOCK), I created the new type (BPF_PROG_TYPE_SOCK_OPS) beccause the existing type expects to be called only once during the connections's lifetime. In contrast, the new program type will be called multiple times from different places in the network stack code. For example, before sending SYN and SYN-ACKs to set an appropriate timeout, when the connection is established to set congestion control, etc. As a result it has "op" field to specify the type of operation requested. The purpose of this new program type is to simplify setting connection parameters, such as buffer sizes, TCP's SYN RTO, etc. For example, it is easy to use facebook's internal IPv6 addresses to determine if both hosts of a connection are in the same datacenter. Therefore, it is easy to write a BPF program to choose a small SYN RTO value when both hosts are in the same datacenter. This patch only contains the framework to support the new BPF program type, following patches add the functionality to set various connection parameters. This patch defines a new BPF program type: BPF_PROG_TYPE_SOCKET_OPS and a new bpf syscall command to load a new program of this type: BPF_PROG_LOAD_SOCKET_OPS. Two new corresponding structs (one for the kernel one for the user/BPF program): /* kernel version */ struct bpf_sock_ops_kern { struct sock *sk; __u32 op; union { __u32 reply; __u32 replylong[4]; }; }; /* user version * Some fields are in network byte order reflecting the sock struct * Use the bpf_ntohl helper macro in samples/bpf/bpf_endian.h to * convert them to host byte order. */ struct bpf_sock_ops { __u32 op; union { __u32 reply; __u32 replylong[4]; }; __u32 family; __u32 remote_ip4; /* In network byte order */ __u32 local_ip4; /* In network byte order */ __u32 remote_ip6[4]; /* In network byte order */ __u32 local_ip6[4]; /* In network byte order */ __u32 remote_port; /* In network byte order */ __u32 local_port; /* In host byte horder */ }; Currently there are two types of ops. The first type expects the BPF program to return a value which is then used by the caller (or a negative value to indicate the operation is not supported). The second type expects state changes to be done by the BPF program, for example through a setsockopt BPF helper function, and they ignore the return value. The reply fields of the bpf_sockt_ops struct are there in case a bpf program needs to return a value larger than an integer. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Daniel Borkmann <daniel@iogearbox.net> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-07-01 11:02:40 +08:00
#else
static inline int tcp_call_bpf(struct sock *sk, int op, u32 nargs, u32 *args)
bpf: BPF support for sock_ops Created a new BPF program type, BPF_PROG_TYPE_SOCK_OPS, and a corresponding struct that allows BPF programs of this type to access some of the socket's fields (such as IP addresses, ports, etc.). It uses the existing bpf cgroups infrastructure so the programs can be attached per cgroup with full inheritance support. The program will be called at appropriate times to set relevant connections parameters such as buffer sizes, SYN and SYN-ACK RTOs, etc., based on connection information such as IP addresses, port numbers, etc. Alghough there are already 3 mechanisms to set parameters (sysctls, route metrics and setsockopts), this new mechanism provides some distinct advantages. Unlike sysctls, it can set parameters per connection. In contrast to route metrics, it can also use port numbers and information provided by a user level program. In addition, it could set parameters probabilistically for evaluation purposes (i.e. do something different on 10% of the flows and compare results with the other 90% of the flows). Also, in cases where IPv6 addresses contain geographic information, the rules to make changes based on the distance (or RTT) between the hosts are much easier than route metric rules and can be global. Finally, unlike setsockopt, it oes not require application changes and it can be updated easily at any time. Although the bpf cgroup framework already contains a sock related program type (BPF_PROG_TYPE_CGROUP_SOCK), I created the new type (BPF_PROG_TYPE_SOCK_OPS) beccause the existing type expects to be called only once during the connections's lifetime. In contrast, the new program type will be called multiple times from different places in the network stack code. For example, before sending SYN and SYN-ACKs to set an appropriate timeout, when the connection is established to set congestion control, etc. As a result it has "op" field to specify the type of operation requested. The purpose of this new program type is to simplify setting connection parameters, such as buffer sizes, TCP's SYN RTO, etc. For example, it is easy to use facebook's internal IPv6 addresses to determine if both hosts of a connection are in the same datacenter. Therefore, it is easy to write a BPF program to choose a small SYN RTO value when both hosts are in the same datacenter. This patch only contains the framework to support the new BPF program type, following patches add the functionality to set various connection parameters. This patch defines a new BPF program type: BPF_PROG_TYPE_SOCKET_OPS and a new bpf syscall command to load a new program of this type: BPF_PROG_LOAD_SOCKET_OPS. Two new corresponding structs (one for the kernel one for the user/BPF program): /* kernel version */ struct bpf_sock_ops_kern { struct sock *sk; __u32 op; union { __u32 reply; __u32 replylong[4]; }; }; /* user version * Some fields are in network byte order reflecting the sock struct * Use the bpf_ntohl helper macro in samples/bpf/bpf_endian.h to * convert them to host byte order. */ struct bpf_sock_ops { __u32 op; union { __u32 reply; __u32 replylong[4]; }; __u32 family; __u32 remote_ip4; /* In network byte order */ __u32 local_ip4; /* In network byte order */ __u32 remote_ip6[4]; /* In network byte order */ __u32 local_ip6[4]; /* In network byte order */ __u32 remote_port; /* In network byte order */ __u32 local_port; /* In host byte horder */ }; Currently there are two types of ops. The first type expects the BPF program to return a value which is then used by the caller (or a negative value to indicate the operation is not supported). The second type expects state changes to be done by the BPF program, for example through a setsockopt BPF helper function, and they ignore the return value. The reply fields of the bpf_sockt_ops struct are there in case a bpf program needs to return a value larger than an integer. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Daniel Borkmann <daniel@iogearbox.net> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-07-01 11:02:40 +08:00
{
return -EPERM;
}
static inline int tcp_call_bpf_2arg(struct sock *sk, int op, u32 arg1, u32 arg2)
{
return -EPERM;
}
static inline int tcp_call_bpf_3arg(struct sock *sk, int op, u32 arg1, u32 arg2,
u32 arg3)
{
return -EPERM;
}
bpf: BPF support for sock_ops Created a new BPF program type, BPF_PROG_TYPE_SOCK_OPS, and a corresponding struct that allows BPF programs of this type to access some of the socket's fields (such as IP addresses, ports, etc.). It uses the existing bpf cgroups infrastructure so the programs can be attached per cgroup with full inheritance support. The program will be called at appropriate times to set relevant connections parameters such as buffer sizes, SYN and SYN-ACK RTOs, etc., based on connection information such as IP addresses, port numbers, etc. Alghough there are already 3 mechanisms to set parameters (sysctls, route metrics and setsockopts), this new mechanism provides some distinct advantages. Unlike sysctls, it can set parameters per connection. In contrast to route metrics, it can also use port numbers and information provided by a user level program. In addition, it could set parameters probabilistically for evaluation purposes (i.e. do something different on 10% of the flows and compare results with the other 90% of the flows). Also, in cases where IPv6 addresses contain geographic information, the rules to make changes based on the distance (or RTT) between the hosts are much easier than route metric rules and can be global. Finally, unlike setsockopt, it oes not require application changes and it can be updated easily at any time. Although the bpf cgroup framework already contains a sock related program type (BPF_PROG_TYPE_CGROUP_SOCK), I created the new type (BPF_PROG_TYPE_SOCK_OPS) beccause the existing type expects to be called only once during the connections's lifetime. In contrast, the new program type will be called multiple times from different places in the network stack code. For example, before sending SYN and SYN-ACKs to set an appropriate timeout, when the connection is established to set congestion control, etc. As a result it has "op" field to specify the type of operation requested. The purpose of this new program type is to simplify setting connection parameters, such as buffer sizes, TCP's SYN RTO, etc. For example, it is easy to use facebook's internal IPv6 addresses to determine if both hosts of a connection are in the same datacenter. Therefore, it is easy to write a BPF program to choose a small SYN RTO value when both hosts are in the same datacenter. This patch only contains the framework to support the new BPF program type, following patches add the functionality to set various connection parameters. This patch defines a new BPF program type: BPF_PROG_TYPE_SOCKET_OPS and a new bpf syscall command to load a new program of this type: BPF_PROG_LOAD_SOCKET_OPS. Two new corresponding structs (one for the kernel one for the user/BPF program): /* kernel version */ struct bpf_sock_ops_kern { struct sock *sk; __u32 op; union { __u32 reply; __u32 replylong[4]; }; }; /* user version * Some fields are in network byte order reflecting the sock struct * Use the bpf_ntohl helper macro in samples/bpf/bpf_endian.h to * convert them to host byte order. */ struct bpf_sock_ops { __u32 op; union { __u32 reply; __u32 replylong[4]; }; __u32 family; __u32 remote_ip4; /* In network byte order */ __u32 local_ip4; /* In network byte order */ __u32 remote_ip6[4]; /* In network byte order */ __u32 local_ip6[4]; /* In network byte order */ __u32 remote_port; /* In network byte order */ __u32 local_port; /* In host byte horder */ }; Currently there are two types of ops. The first type expects the BPF program to return a value which is then used by the caller (or a negative value to indicate the operation is not supported). The second type expects state changes to be done by the BPF program, for example through a setsockopt BPF helper function, and they ignore the return value. The reply fields of the bpf_sockt_ops struct are there in case a bpf program needs to return a value larger than an integer. Signed-off-by: Lawrence Brakmo <brakmo@fb.com> Acked-by: Daniel Borkmann <daniel@iogearbox.net> Acked-by: Alexei Starovoitov <ast@kernel.org> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-07-01 11:02:40 +08:00
#endif
static inline u32 tcp_timeout_init(struct sock *sk)
{
int timeout;
timeout = tcp_call_bpf(sk, BPF_SOCK_OPS_TIMEOUT_INIT, 0, NULL);
if (timeout <= 0)
timeout = TCP_TIMEOUT_INIT;
return timeout;
}
static inline u32 tcp_rwnd_init_bpf(struct sock *sk)
{
int rwnd;
rwnd = tcp_call_bpf(sk, BPF_SOCK_OPS_RWND_INIT, 0, NULL);
if (rwnd < 0)
rwnd = 0;
return rwnd;
}
static inline bool tcp_bpf_ca_needs_ecn(struct sock *sk)
{
return (tcp_call_bpf(sk, BPF_SOCK_OPS_NEEDS_ECN, 0, NULL) == 1);
}
static inline void tcp_bpf_rtt(struct sock *sk)
{
if (BPF_SOCK_OPS_TEST_FLAG(tcp_sk(sk), BPF_SOCK_OPS_RTT_CB_FLAG))
tcp_call_bpf(sk, BPF_SOCK_OPS_RTT_CB, 0, NULL);
}
#if IS_ENABLED(CONFIG_SMC)
extern struct static_key_false tcp_have_smc;
#endif
#if IS_ENABLED(CONFIG_TLS_DEVICE)
void clean_acked_data_enable(struct inet_connection_sock *icsk,
void (*cad)(struct sock *sk, u32 ack_seq));
void clean_acked_data_disable(struct inet_connection_sock *icsk);
void clean_acked_data_flush(void);
#endif
tcp: add optional per socket transmit delay Adding delays to TCP flows is crucial for studying behavior of TCP stacks, including congestion control modules. Linux offers netem module, but it has unpractical constraints : - Need root access to change qdisc - Hard to setup on egress if combined with non trivial qdisc like FQ - Single delay for all flows. EDT (Earliest Departure Time) adoption in TCP stack allows us to enable a per socket delay at a very small cost. Networking tools can now establish thousands of flows, each of them with a different delay, simulating real world conditions. This requires FQ packet scheduler or a EDT-enabled NIC. This patchs adds TCP_TX_DELAY socket option, to set a delay in usec units. unsigned int tx_delay = 10000; /* 10 msec */ setsockopt(fd, SOL_TCP, TCP_TX_DELAY, &tx_delay, sizeof(tx_delay)); Note that FQ packet scheduler limits might need some tweaking : man tc-fq PARAMETERS limit Hard limit on the real queue size. When this limit is reached, new packets are dropped. If the value is lowered, packets are dropped so that the new limit is met. Default is 10000 packets. flow_limit Hard limit on the maximum number of packets queued per flow. Default value is 100. Use of TCP_TX_DELAY option will increase number of skbs in FQ qdisc, so packets would be dropped if any of the previous limit is hit. Use of a jump label makes this support runtime-free, for hosts never using the option. Also note that TSQ (TCP Small Queues) limits are slightly changed with this patch : we need to account that skbs artificially delayed wont stop us providind more skbs to feed the pipe (netem uses skb_orphan_partial() for this purpose, but FQ can not use this trick) Because of that, using big delays might very well trigger old bugs in TSO auto defer logic and/or sndbuf limited detection. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2019-06-13 02:57:25 +08:00
DECLARE_STATIC_KEY_FALSE(tcp_tx_delay_enabled);
static inline void tcp_add_tx_delay(struct sk_buff *skb,
const struct tcp_sock *tp)
{
if (static_branch_unlikely(&tcp_tx_delay_enabled))
skb->skb_mstamp_ns += (u64)tp->tcp_tx_delay * NSEC_PER_USEC;
}
/* Compute Earliest Departure Time for some control packets
* like ACK or RST for TIME_WAIT or non ESTABLISHED sockets.
*/
static inline u64 tcp_transmit_time(const struct sock *sk)
tcp: add optional per socket transmit delay Adding delays to TCP flows is crucial for studying behavior of TCP stacks, including congestion control modules. Linux offers netem module, but it has unpractical constraints : - Need root access to change qdisc - Hard to setup on egress if combined with non trivial qdisc like FQ - Single delay for all flows. EDT (Earliest Departure Time) adoption in TCP stack allows us to enable a per socket delay at a very small cost. Networking tools can now establish thousands of flows, each of them with a different delay, simulating real world conditions. This requires FQ packet scheduler or a EDT-enabled NIC. This patchs adds TCP_TX_DELAY socket option, to set a delay in usec units. unsigned int tx_delay = 10000; /* 10 msec */ setsockopt(fd, SOL_TCP, TCP_TX_DELAY, &tx_delay, sizeof(tx_delay)); Note that FQ packet scheduler limits might need some tweaking : man tc-fq PARAMETERS limit Hard limit on the real queue size. When this limit is reached, new packets are dropped. If the value is lowered, packets are dropped so that the new limit is met. Default is 10000 packets. flow_limit Hard limit on the maximum number of packets queued per flow. Default value is 100. Use of TCP_TX_DELAY option will increase number of skbs in FQ qdisc, so packets would be dropped if any of the previous limit is hit. Use of a jump label makes this support runtime-free, for hosts never using the option. Also note that TSQ (TCP Small Queues) limits are slightly changed with this patch : we need to account that skbs artificially delayed wont stop us providind more skbs to feed the pipe (netem uses skb_orphan_partial() for this purpose, but FQ can not use this trick) Because of that, using big delays might very well trigger old bugs in TSO auto defer logic and/or sndbuf limited detection. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2019-06-13 02:57:25 +08:00
{
if (static_branch_unlikely(&tcp_tx_delay_enabled)) {
u32 delay = (sk->sk_state == TCP_TIME_WAIT) ?
tcp_twsk(sk)->tw_tx_delay : tcp_sk(sk)->tcp_tx_delay;
return tcp_clock_ns() + (u64)delay * NSEC_PER_USEC;
tcp: add optional per socket transmit delay Adding delays to TCP flows is crucial for studying behavior of TCP stacks, including congestion control modules. Linux offers netem module, but it has unpractical constraints : - Need root access to change qdisc - Hard to setup on egress if combined with non trivial qdisc like FQ - Single delay for all flows. EDT (Earliest Departure Time) adoption in TCP stack allows us to enable a per socket delay at a very small cost. Networking tools can now establish thousands of flows, each of them with a different delay, simulating real world conditions. This requires FQ packet scheduler or a EDT-enabled NIC. This patchs adds TCP_TX_DELAY socket option, to set a delay in usec units. unsigned int tx_delay = 10000; /* 10 msec */ setsockopt(fd, SOL_TCP, TCP_TX_DELAY, &tx_delay, sizeof(tx_delay)); Note that FQ packet scheduler limits might need some tweaking : man tc-fq PARAMETERS limit Hard limit on the real queue size. When this limit is reached, new packets are dropped. If the value is lowered, packets are dropped so that the new limit is met. Default is 10000 packets. flow_limit Hard limit on the maximum number of packets queued per flow. Default value is 100. Use of TCP_TX_DELAY option will increase number of skbs in FQ qdisc, so packets would be dropped if any of the previous limit is hit. Use of a jump label makes this support runtime-free, for hosts never using the option. Also note that TSQ (TCP Small Queues) limits are slightly changed with this patch : we need to account that skbs artificially delayed wont stop us providind more skbs to feed the pipe (netem uses skb_orphan_partial() for this purpose, but FQ can not use this trick) Because of that, using big delays might very well trigger old bugs in TSO auto defer logic and/or sndbuf limited detection. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2019-06-13 02:57:25 +08:00
}
return 0;
tcp: add optional per socket transmit delay Adding delays to TCP flows is crucial for studying behavior of TCP stacks, including congestion control modules. Linux offers netem module, but it has unpractical constraints : - Need root access to change qdisc - Hard to setup on egress if combined with non trivial qdisc like FQ - Single delay for all flows. EDT (Earliest Departure Time) adoption in TCP stack allows us to enable a per socket delay at a very small cost. Networking tools can now establish thousands of flows, each of them with a different delay, simulating real world conditions. This requires FQ packet scheduler or a EDT-enabled NIC. This patchs adds TCP_TX_DELAY socket option, to set a delay in usec units. unsigned int tx_delay = 10000; /* 10 msec */ setsockopt(fd, SOL_TCP, TCP_TX_DELAY, &tx_delay, sizeof(tx_delay)); Note that FQ packet scheduler limits might need some tweaking : man tc-fq PARAMETERS limit Hard limit on the real queue size. When this limit is reached, new packets are dropped. If the value is lowered, packets are dropped so that the new limit is met. Default is 10000 packets. flow_limit Hard limit on the maximum number of packets queued per flow. Default value is 100. Use of TCP_TX_DELAY option will increase number of skbs in FQ qdisc, so packets would be dropped if any of the previous limit is hit. Use of a jump label makes this support runtime-free, for hosts never using the option. Also note that TSQ (TCP Small Queues) limits are slightly changed with this patch : we need to account that skbs artificially delayed wont stop us providind more skbs to feed the pipe (netem uses skb_orphan_partial() for this purpose, but FQ can not use this trick) Because of that, using big delays might very well trigger old bugs in TSO auto defer logic and/or sndbuf limited detection. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2019-06-13 02:57:25 +08:00
}
#endif /* _TCP_H */