2019-05-27 08:55:01 +02:00
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2005-04-16 15:20:36 -07:00
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2005-05-05 16:16:16 -07:00
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2005-04-16 15:20:36 -07:00
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2011-11-23 20:12:59 -05:00
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2005-04-16 15:20:36 -07:00
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2005-11-10 13:01:24 -08:00
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TCPCT part 1d: define TCP cookie option, extend existing struct's
Data structures are carefully composed to require minimal additions.
For example, the struct tcp_options_received cookie_plus variable fits
between existing 16-bit and 8-bit variables, requiring no additional
space (taking alignment into consideration). There are no additions to
tcp_request_sock, and only 1 pointer in tcp_sock.
This is a significantly revised implementation of an earlier (year-old)
patch that no longer applies cleanly, with permission of the original
author (Adam Langley):
http://thread.gmane.org/gmane.linux.network/102586
The principle difference is using a TCP option to carry the cookie nonce,
instead of a user configured offset in the data. This is more flexible and
less subject to user configuration error. Such a cookie option has been
suggested for many years, and is also useful without SYN data, allowing
several related concepts to use the same extension option.
"Re: SYN floods (was: does history repeat itself?)", September 9, 1996.
http://www.merit.net/mail.archives/nanog/1996-09/msg00235.html
"Re: what a new TCP header might look like", May 12, 1998.
ftp://ftp.isi.edu/end2end/end2end-interest-1998.mail
These functions will also be used in subsequent patches that implement
additional features.
Requires:
TCPCT part 1a: add request_values parameter for sending SYNACK
TCPCT part 1b: generate Responder Cookie secret
TCPCT part 1c: sysctl_tcp_cookie_size, socket option TCP_COOKIE_TRANSACTIONS
Signed-off-by: William.Allen.Simpson@gmail.com
Signed-off-by: David S. Miller <davem@davemloft.net>
2009-12-02 18:17:05 +00:00
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2014-02-26 14:02:48 -08:00
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2020-06-19 12:12:34 -07:00
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2025-03-05 23:38:43 +01:00
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2005-08-09 20:11:08 -07:00
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2005-08-09 20:44:40 -07:00
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2005-08-09 20:00:51 -07:00
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2005-04-16 15:20:36 -07:00
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[NET] Generalise TCP's struct open_request minisock infrastructure
Kept this first changeset minimal, without changing existing names to
ease peer review.
Basicaly tcp_openreq_alloc now receives the or_calltable, that in turn
has two new members:
->slab, that replaces tcp_openreq_cachep
->obj_size, to inform the size of the openreq descendant for
a specific protocol
The protocol specific fields in struct open_request were moved to a
class hierarchy, with the things that are common to all connection
oriented PF_INET protocols in struct inet_request_sock, the TCP ones
in tcp_request_sock, that is an inet_request_sock, that is an
open_request.
I.e. this uses the same approach used for the struct sock class
hierarchy, with sk_prot indicating if the protocol wants to use the
open_request infrastructure by filling in sk_prot->rsk_prot with an
or_calltable.
Results? Performance is improved and TCP v4 now uses only 64 bytes per
open request minisock, down from 96 without this patch :-)
Next changeset will rename some of the structs, fields and functions
mentioned above, struct or_calltable is way unclear, better name it
struct request_sock_ops, s/struct open_request/struct request_sock/g,
etc.
Signed-off-by: Arnaldo Carvalho de Melo <acme@ghostprotocols.net>
Signed-off-by: David S. Miller <davem@davemloft.net>
2005-06-18 22:46:52 -07:00
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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 01:01:21 -07:00
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2005-04-16 15:20:36 -07:00
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2005-08-09 20:08:28 -07:00
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2023-10-23 20:21:54 +01:00
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2007-05-27 02:04:16 -07:00
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2009-05-04 11:11:01 -07:00
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2020-01-09 07:59:20 -08:00
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tcp: drop secpath at the same time as we currently drop dst
Xiumei reported hitting the WARN in xfrm6_tunnel_net_exit while
running tests that boil down to:
- create a pair of netns
- run a basic TCP test over ipcomp6
- delete the pair of netns
The xfrm_state found on spi_byaddr was not deleted at the time we
delete the netns, because we still have a reference on it. This
lingering reference comes from a secpath (which holds a ref on the
xfrm_state), which is still attached to an skb. This skb is not
leaked, it ends up on sk_receive_queue and then gets defer-free'd by
skb_attempt_defer_free.
The problem happens when we defer freeing an skb (push it on one CPU's
defer_list), and don't flush that list before the netns is deleted. In
that case, we still have a reference on the xfrm_state that we don't
expect at this point.
We already drop the skb's dst in the TCP receive path when it's no
longer needed, so let's also drop the secpath. At this point,
tcp_filter has already called into the LSM hooks that may require the
secpath, so it should not be needed anymore. However, in some of those
places, the MPTCP extension has just been attached to the skb, so we
cannot simply drop all extensions.
Fixes: 68822bdf76f1 ("net: generalize skb freeing deferral to per-cpu lists")
Reported-by: Xiumei Mu <xmu@redhat.com>
Signed-off-by: Sabrina Dubroca <sd@queasysnail.net>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/5055ba8f8f72bdcb602faa299faca73c280b7735.1739743613.git.sd@queasysnail.net
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-02-17 11:23:35 +01:00
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2026-03-02 20:55:27 +00:00
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2005-08-09 20:08:28 -07:00
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2005-04-16 15:20:36 -07:00
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2011-12-11 21:47:02 +00:00
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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>
2017-06-30 20:02:40 -07:00
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2019-06-19 23:46:28 +02:00
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|
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>
2017-06-30 20:02:40 -07:00
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2005-08-09 20:07:35 -07:00
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2005-04-16 15:20:36 -07:00
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2021-10-14 06:41:26 -07:00
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2025-08-29 21:56:38 +00:00
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2013-09-23 11:33:32 -07:00
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2005-04-16 15:20:36 -07:00
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2020-04-17 07:10:23 -07:00
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2008-07-19 00:04:31 -07:00
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2019-05-17 17:17:22 -07:00
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2005-04-16 15:20:36 -07:00
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2014-10-20 18:15:50 +09:00
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2005-04-16 15:20:36 -07:00
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2014-10-20 18:15:50 +09:00
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2005-04-16 15:20:36 -07:00
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2019-08-07 19:52:30 -04:00
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2015-03-06 11:18:22 +08:00
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2006-03-20 17:53:41 -08:00
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2015-03-06 11:18:24 +08:00
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2015-03-06 11:18:23 +08:00
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2005-04-16 15:20:36 -07:00
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2017-04-04 21:09:48 +08:00
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2025-09-16 10:24:31 +02:00
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2005-04-16 15:20:36 -07:00
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2012-08-31 02:48:31 +00:00
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2005-04-16 15:20:36 -07:00
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2012-08-31 02:48:31 +00:00
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2005-04-16 15:20:36 -07:00
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2020-04-24 16:06:16 +08:00
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2005-04-16 15:20:36 -07:00
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2023-10-06 01:18:40 +00:00
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2005-04-16 15:20:36 -07:00
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2025-02-07 15:28:29 +00:00
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2017-07-19 15:41:26 -07:00
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2023-10-15 13:47:00 -04:00
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2012-04-12 19:48:40 +00:00
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2011-06-08 11:08:38 +00:00
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2005-04-16 15:20:36 -07:00
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2023-10-20 12:57:46 +00:00
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2005-04-16 15:20:36 -07:00
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2014-10-20 18:15:50 +09:00
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2005-04-16 15:20:36 -07:00
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2006-11-14 19:07:45 -08:00
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2023-10-23 20:21:58 +01:00
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2020-01-09 07:59:17 -08:00
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2015-04-06 14:37:26 -07:00
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tcp: accecn: AccECN option
The Accurate ECN allows echoing back the sum of bytes for
each IP ECN field value in the received packets using
AccECN option. This change implements AccECN option tx & rx
side processing without option send control related features
that are added by a later change.
Based on specification:
https://tools.ietf.org/id/draft-ietf-tcpm-accurate-ecn-28.txt
(Some features of the spec will be added in the later changes
rather than in this one).
A full-length AccECN option is always attempted but if it does
not fit, the minimum length is selected based on the counters
that have changed since the last update. The AccECN option
(with 24-bit fields) often ends in odd sizes so the option
write code tries to take advantage of some nop used to pad
the other TCP options.
The delivered_ecn_bytes pairs with received_ecn_bytes similar
to how delivered_ce pairs with received_ce. In contrast to
ACE field, however, the option is not always available to update
delivered_ecn_bytes. For ACK w/o AccECN option, the delivered
bytes calculated based on the cumulative ACK+SACK information
are assigned to one of the counters using an estimation
heuristic to select the most likely ECN byte counter. Any
estimation error is corrected when the next AccECN option
arrives. It may occur that the heuristic gets too confused
when there are enough different byte counter deltas between
ACKs with the AccECN option in which case the heuristic just
gives up on updating the counters for a while.
tcp_ecn_option sysctl can be used to select option sending
mode for AccECN: TCP_ECN_OPTION_DISABLED, TCP_ECN_OPTION_MINIMUM,
and TCP_ECN_OPTION_FULL.
This patch increases the size of tcp_info struct, as there is
no existing holes for new u32 variables. Below are the pahole
outcomes before and after this patch:
[BEFORE THIS PATCH]
struct tcp_info {
[...]
__u32 tcpi_total_rto_time; /* 244 4 */
/* size: 248, cachelines: 4, members: 61 */
}
[AFTER THIS PATCH]
struct tcp_info {
[...]
__u32 tcpi_total_rto_time; /* 244 4 */
__u32 tcpi_received_ce; /* 248 4 */
__u32 tcpi_delivered_e1_bytes; /* 252 4 */
__u32 tcpi_delivered_e0_bytes; /* 256 4 */
__u32 tcpi_delivered_ce_bytes; /* 260 4 */
__u32 tcpi_received_e1_bytes; /* 264 4 */
__u32 tcpi_received_e0_bytes; /* 268 4 */
__u32 tcpi_received_ce_bytes; /* 272 4 */
/* size: 280, cachelines: 5, members: 68 */
}
This patch uses the existing 1-byte holes in the tcp_sock_write_txrx
group for new u8 members, but adds a 4-byte hole in tcp_sock_write_rx
group after the new u32 delivered_ecn_bytes[3] member. Therefore, the
group size of tcp_sock_write_rx is increased from 96 to 112. Below
are the pahole outcomes before and after this patch:
[BEFORE THIS PATCH]
struct tcp_sock {
[...]
u8 received_ce_pending:4; /* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
/* XXX 1 byte hole, try to pack */
[...]
u32 rcv_rtt_last_tsecr; /* 2668 4 */
[...]
__cacheline_group_end__tcp_sock_write_rx[0]; /* 2728 0 */
[...]
/* size: 3200, cachelines: 50, members: 167 */
}
[AFTER THIS PATCH]
struct tcp_sock {
[...]
u8 received_ce_pending:4;/* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
u8 accecn_minlen:2; /* 2523: 0 1 */
u8 est_ecnfield:2; /* 2523: 2 1 */
u8 unused3:4; /* 2523: 4 1 */
[...]
u32 rcv_rtt_last_tsecr; /* 2668 4 */
u32 delivered_ecn_bytes[3];/* 2672 12 */
/* XXX 4 bytes hole, try to pack */
[...]
__cacheline_group_end__tcp_sock_write_rx[0]; /* 2744 0 */
[...]
/* size: 3200, cachelines: 50, members: 171 */
}
Signed-off-by: Ilpo Järvinen <ij@kernel.org>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Co-developed-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Signed-off-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250916082434.100722-7-chia-yu.chang@nokia-bell-labs.com
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-09-16 10:24:30 +02:00
|
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|
2012-07-19 06:43:05 +00:00
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2017-10-25 11:01:45 +02:00
|
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2005-04-16 15:20:36 -07:00
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2006-11-14 19:07:45 -08:00
|
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|
2015-04-06 14:37:26 -07:00
|
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|
|
tcp: accecn: AccECN option
The Accurate ECN allows echoing back the sum of bytes for
each IP ECN field value in the received packets using
AccECN option. This change implements AccECN option tx & rx
side processing without option send control related features
that are added by a later change.
Based on specification:
https://tools.ietf.org/id/draft-ietf-tcpm-accurate-ecn-28.txt
(Some features of the spec will be added in the later changes
rather than in this one).
A full-length AccECN option is always attempted but if it does
not fit, the minimum length is selected based on the counters
that have changed since the last update. The AccECN option
(with 24-bit fields) often ends in odd sizes so the option
write code tries to take advantage of some nop used to pad
the other TCP options.
The delivered_ecn_bytes pairs with received_ecn_bytes similar
to how delivered_ce pairs with received_ce. In contrast to
ACE field, however, the option is not always available to update
delivered_ecn_bytes. For ACK w/o AccECN option, the delivered
bytes calculated based on the cumulative ACK+SACK information
are assigned to one of the counters using an estimation
heuristic to select the most likely ECN byte counter. Any
estimation error is corrected when the next AccECN option
arrives. It may occur that the heuristic gets too confused
when there are enough different byte counter deltas between
ACKs with the AccECN option in which case the heuristic just
gives up on updating the counters for a while.
tcp_ecn_option sysctl can be used to select option sending
mode for AccECN: TCP_ECN_OPTION_DISABLED, TCP_ECN_OPTION_MINIMUM,
and TCP_ECN_OPTION_FULL.
This patch increases the size of tcp_info struct, as there is
no existing holes for new u32 variables. Below are the pahole
outcomes before and after this patch:
[BEFORE THIS PATCH]
struct tcp_info {
[...]
__u32 tcpi_total_rto_time; /* 244 4 */
/* size: 248, cachelines: 4, members: 61 */
}
[AFTER THIS PATCH]
struct tcp_info {
[...]
__u32 tcpi_total_rto_time; /* 244 4 */
__u32 tcpi_received_ce; /* 248 4 */
__u32 tcpi_delivered_e1_bytes; /* 252 4 */
__u32 tcpi_delivered_e0_bytes; /* 256 4 */
__u32 tcpi_delivered_ce_bytes; /* 260 4 */
__u32 tcpi_received_e1_bytes; /* 264 4 */
__u32 tcpi_received_e0_bytes; /* 268 4 */
__u32 tcpi_received_ce_bytes; /* 272 4 */
/* size: 280, cachelines: 5, members: 68 */
}
This patch uses the existing 1-byte holes in the tcp_sock_write_txrx
group for new u8 members, but adds a 4-byte hole in tcp_sock_write_rx
group after the new u32 delivered_ecn_bytes[3] member. Therefore, the
group size of tcp_sock_write_rx is increased from 96 to 112. Below
are the pahole outcomes before and after this patch:
[BEFORE THIS PATCH]
struct tcp_sock {
[...]
u8 received_ce_pending:4; /* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
/* XXX 1 byte hole, try to pack */
[...]
u32 rcv_rtt_last_tsecr; /* 2668 4 */
[...]
__cacheline_group_end__tcp_sock_write_rx[0]; /* 2728 0 */
[...]
/* size: 3200, cachelines: 50, members: 167 */
}
[AFTER THIS PATCH]
struct tcp_sock {
[...]
u8 received_ce_pending:4;/* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
u8 accecn_minlen:2; /* 2523: 0 1 */
u8 est_ecnfield:2; /* 2523: 2 1 */
u8 unused3:4; /* 2523: 4 1 */
[...]
u32 rcv_rtt_last_tsecr; /* 2668 4 */
u32 delivered_ecn_bytes[3];/* 2672 12 */
/* XXX 4 bytes hole, try to pack */
[...]
__cacheline_group_end__tcp_sock_write_rx[0]; /* 2744 0 */
[...]
/* size: 3200, cachelines: 50, members: 171 */
}
Signed-off-by: Ilpo Järvinen <ij@kernel.org>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Co-developed-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Signed-off-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250916082434.100722-7-chia-yu.chang@nokia-bell-labs.com
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-09-16 10:24:30 +02:00
|
|
|
|
2012-07-19 06:43:05 +00:00
|
|
|
|
2017-10-25 11:01:45 +02:00
|
|
|
|
2005-04-16 15:20:36 -07:00
|
|
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2006-11-14 19:07:45 -08:00
|
|
|
|
2008-07-19 00:04:31 -07:00
|
|
|
|
2017-10-25 11:01:45 +02:00
|
|
|
|
tcp: accecn: AccECN option
The Accurate ECN allows echoing back the sum of bytes for
each IP ECN field value in the received packets using
AccECN option. This change implements AccECN option tx & rx
side processing without option send control related features
that are added by a later change.
Based on specification:
https://tools.ietf.org/id/draft-ietf-tcpm-accurate-ecn-28.txt
(Some features of the spec will be added in the later changes
rather than in this one).
A full-length AccECN option is always attempted but if it does
not fit, the minimum length is selected based on the counters
that have changed since the last update. The AccECN option
(with 24-bit fields) often ends in odd sizes so the option
write code tries to take advantage of some nop used to pad
the other TCP options.
The delivered_ecn_bytes pairs with received_ecn_bytes similar
to how delivered_ce pairs with received_ce. In contrast to
ACE field, however, the option is not always available to update
delivered_ecn_bytes. For ACK w/o AccECN option, the delivered
bytes calculated based on the cumulative ACK+SACK information
are assigned to one of the counters using an estimation
heuristic to select the most likely ECN byte counter. Any
estimation error is corrected when the next AccECN option
arrives. It may occur that the heuristic gets too confused
when there are enough different byte counter deltas between
ACKs with the AccECN option in which case the heuristic just
gives up on updating the counters for a while.
tcp_ecn_option sysctl can be used to select option sending
mode for AccECN: TCP_ECN_OPTION_DISABLED, TCP_ECN_OPTION_MINIMUM,
and TCP_ECN_OPTION_FULL.
This patch increases the size of tcp_info struct, as there is
no existing holes for new u32 variables. Below are the pahole
outcomes before and after this patch:
[BEFORE THIS PATCH]
struct tcp_info {
[...]
__u32 tcpi_total_rto_time; /* 244 4 */
/* size: 248, cachelines: 4, members: 61 */
}
[AFTER THIS PATCH]
struct tcp_info {
[...]
__u32 tcpi_total_rto_time; /* 244 4 */
__u32 tcpi_received_ce; /* 248 4 */
__u32 tcpi_delivered_e1_bytes; /* 252 4 */
__u32 tcpi_delivered_e0_bytes; /* 256 4 */
__u32 tcpi_delivered_ce_bytes; /* 260 4 */
__u32 tcpi_received_e1_bytes; /* 264 4 */
__u32 tcpi_received_e0_bytes; /* 268 4 */
__u32 tcpi_received_ce_bytes; /* 272 4 */
/* size: 280, cachelines: 5, members: 68 */
}
This patch uses the existing 1-byte holes in the tcp_sock_write_txrx
group for new u8 members, but adds a 4-byte hole in tcp_sock_write_rx
group after the new u32 delivered_ecn_bytes[3] member. Therefore, the
group size of tcp_sock_write_rx is increased from 96 to 112. Below
are the pahole outcomes before and after this patch:
[BEFORE THIS PATCH]
struct tcp_sock {
[...]
u8 received_ce_pending:4; /* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
/* XXX 1 byte hole, try to pack */
[...]
u32 rcv_rtt_last_tsecr; /* 2668 4 */
[...]
__cacheline_group_end__tcp_sock_write_rx[0]; /* 2728 0 */
[...]
/* size: 3200, cachelines: 50, members: 167 */
}
[AFTER THIS PATCH]
struct tcp_sock {
[...]
u8 received_ce_pending:4;/* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
u8 accecn_minlen:2; /* 2523: 0 1 */
u8 est_ecnfield:2; /* 2523: 2 1 */
u8 unused3:4; /* 2523: 4 1 */
[...]
u32 rcv_rtt_last_tsecr; /* 2668 4 */
u32 delivered_ecn_bytes[3];/* 2672 12 */
/* XXX 4 bytes hole, try to pack */
[...]
__cacheline_group_end__tcp_sock_write_rx[0]; /* 2744 0 */
[...]
/* size: 3200, cachelines: 50, members: 171 */
}
Signed-off-by: Ilpo Järvinen <ij@kernel.org>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Co-developed-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Signed-off-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250916082434.100722-7-chia-yu.chang@nokia-bell-labs.com
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-09-16 10:24:30 +02:00
|
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2025-09-16 10:24:33 +02:00
|
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|
2005-04-16 15:20:36 -07:00
|
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2005-11-10 17:13:47 -08:00
|
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|
2005-04-16 15:20:36 -07:00
|
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|
|
2010-02-18 02:47:01 +00:00
|
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2016-01-27 22:54:06 +01:00
|
|
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|
2011-02-02 17:05:11 -08:00
|
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|
2012-07-19 06:43:09 +00:00
|
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2012-08-31 12:29:11 +00:00
|
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|
2012-07-19 06:43:11 +00:00
|
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|
2012-07-19 06:43:09 +00:00
|
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2012-08-31 12:29:11 +00:00
|
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2016-08-22 17:17:54 -07:00
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2012-08-31 12:29:11 +00:00
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2005-08-09 20:44:40 -07:00
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2005-04-16 15:20:36 -07:00
|
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|
2013-10-19 16:25:36 -07:00
|
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|
2017-10-26 21:54:57 -07:00
|
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2017-01-12 22:11:36 -08:00
|
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2017-11-03 16:38:48 -07:00
|
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|
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>
2018-05-16 16:40:10 -07:00
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|
2017-01-12 22:11:36 -08:00
|
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|
2022-06-08 23:34:08 -07:00
|
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|
|
2008-11-25 21:16:35 -08:00
|
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|
2017-06-07 13:29:12 -07:00
|
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2005-04-16 15:20:36 -07:00
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2015-05-15 12:39:27 -07:00
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2025-08-15 20:16:15 +00:00
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2025-08-15 20:16:17 +00:00
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|
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>
2016-01-14 15:21:14 -08:00
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2015-05-15 12:39:27 -07:00
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net: Allow opt-out from global protocol memory accounting.
Some protocols (e.g., TCP, UDP) implement memory accounting for socket
buffers and charge memory to per-protocol global counters pointed to by
sk->sk_proto->memory_allocated.
Sometimes, system processes do not want that limitation. For a similar
purpose, there is SO_RESERVE_MEM for sockets under memcg.
Also, by opting out of the per-protocol accounting, sockets under memcg
can avoid paying costs for two orthogonal memory accounting mechanisms.
A microbenchmark result is in the subsequent bpf patch.
Let's allow opt-out from the per-protocol memory accounting if
sk->sk_bypass_prot_mem is true.
sk->sk_bypass_prot_mem and sk->sk_prot are placed in the same cache
line, and sk_has_account() always fetches sk->sk_prot before accessing
sk->sk_bypass_prot_mem, so there is no extra cache miss for this patch.
The following patches will set sk->sk_bypass_prot_mem to true, and
then, the per-protocol memory accounting will be skipped.
Note that this does NOT disable memcg, but rather the per-protocol one.
Another option not to use the hole in struct sock_common is create
sk_prot variants like tcp_prot_bypass, but this would complicate
SOCKMAP logic, tcp_bpf_prots etc.
Signed-off-by: Kuniyuki Iwashima <kuniyu@google.com>
Signed-off-by: Martin KaFai Lau <martin.lau@kernel.org>
Reviewed-by: Shakeel Butt <shakeel.butt@linux.dev>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Acked-by: Roman Gushchin <roman.gushchin@linux.dev>
Link: https://patch.msgid.link/20251014235604.3057003-3-kuniyu@google.com
2025-10-14 23:54:55 +00:00
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2019-10-09 15:10:15 -07:00
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2015-05-15 12:39:27 -07:00
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2005-04-16 15:20:36 -07:00
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2012-05-16 23:15:34 +00:00
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2005-04-16 15:20:36 -07:00
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2007-01-04 12:25:16 -08:00
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2005-04-16 15:20:36 -07:00
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[TCP]: Fix ambiguity in the `before' relation.
While looking at DCCP sequence numbers, I stumbled over a problem with
the following definition of before in tcp.h:
static inline int before(__u32 seq1, __u32 seq2)
{
return (__s32)(seq1-seq2) < 0;
}
Problem: This definition suffers from an an ambiguity, i.e. always
before(a, (a + 2^31) % 2^32)) = 1
before((a + 2^31) % 2^32), a) = 1
In text: when the difference between a and b amounts to 2^31,
a is always considered `before' b, the function can not decide.
The reason is that implicitly 0 is `before' 1 ... 2^31-1 ... 2^31
Solution: There is a simple fix, by defining before in such a way that
0 is no longer `before' 2^31, i.e. 0 `before' 1 ... 2^31-1
By not using the middle between 0 and 2^32, before can be made
unambiguous.
This is achieved by testing whether seq2-seq1 > 0 (using signed
32-bit arithmetic).
I attach a patch to codify this. Also the `after' relation is basically
a redefinition of `before', it is now defined as a macro after before.
Signed-off-by: Gerrit Renker <gerrit@erg.abdn.ac.uk>
Signed-off-by: David S. Miller <davem@davemloft.net>
2006-12-20 10:25:55 -08:00
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2005-04-16 15:20:36 -07:00
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2012-05-16 23:15:34 +00:00
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2005-04-16 15:20:36 -07:00
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2021-10-29 22:05:41 -04:00
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2021-11-02 22:58:44 -04:00
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2021-10-29 22:05:41 -04:00
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2015-05-15 12:39:26 -07:00
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2024-04-29 13:40:23 +00:00
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2012-01-30 14:16:06 -08:00
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2009-04-19 09:43:48 +00:00
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2005-04-16 15:20:36 -07:00
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2008-07-18 04:02:08 -07:00
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2016-04-27 16:44:43 -07:00
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2008-07-18 04:02:08 -07:00
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2010-04-22 07:00:24 +00:00
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2005-04-16 15:20:36 -07:00
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2026-01-30 20:24:26 +01:00
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2025-06-16 08:10:47 -10:00
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tcp: TCP Small Queues
This introduce TSQ (TCP Small Queues)
TSQ goal is to reduce number of TCP packets in xmit queues (qdisc &
device queues), to reduce RTT and cwnd bias, part of the bufferbloat
problem.
sk->sk_wmem_alloc not allowed to grow above a given limit,
allowing no more than ~128KB [1] per tcp socket in qdisc/dev layers at a
given time.
TSO packets are sized/capped to half the limit, so that we have two
TSO packets in flight, allowing better bandwidth use.
As a side effect, setting the limit to 40000 automatically reduces the
standard gso max limit (65536) to 40000/2 : It can help to reduce
latencies of high prio packets, having smaller TSO packets.
This means we divert sock_wfree() to a tcp_wfree() handler, to
queue/send following frames when skb_orphan() [2] is called for the
already queued skbs.
Results on my dev machines (tg3/ixgbe nics) are really impressive,
using standard pfifo_fast, and with or without TSO/GSO.
Without reduction of nominal bandwidth, we have reduction of buffering
per bulk sender :
< 1ms on Gbit (instead of 50ms with TSO)
< 8ms on 100Mbit (instead of 132 ms)
I no longer have 4 MBytes backlogged in qdisc by a single netperf
session, and both side socket autotuning no longer use 4 Mbytes.
As skb destructor cannot restart xmit itself ( as qdisc lock might be
taken at this point ), we delegate the work to a tasklet. We use one
tasklest per cpu for performance reasons.
If tasklet finds a socket owned by the user, it sets TSQ_OWNED flag.
This flag is tested in a new protocol method called from release_sock(),
to eventually send new segments.
[1] New /proc/sys/net/ipv4/tcp_limit_output_bytes tunable
[2] skb_orphan() is usually called at TX completion time,
but some drivers call it in their start_xmit() handler.
These drivers should at least use BQL, or else a single TCP
session can still fill the whole NIC TX ring, since TSQ will
have no effect.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Dave Taht <dave.taht@bufferbloat.net>
Cc: Tom Herbert <therbert@google.com>
Cc: Matt Mathis <mattmathis@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Cc: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2012-07-11 05:50:31 +00:00
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2018-11-08 12:19:21 +01:00
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2005-04-16 15:20:36 -07:00
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2013-09-23 11:33:32 -07:00
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2005-04-16 15:20:36 -07:00
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2013-09-23 11:33:32 -07:00
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2005-04-16 15:20:36 -07:00
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2021-10-27 13:19:18 -07:00
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2015-03-02 15:37:48 +08:00
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2017-07-28 16:22:41 -07:00
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2022-09-26 16:27:37 -07:00
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2023-06-07 19:19:13 +01:00
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2020-01-09 07:59:21 -08:00
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2023-06-09 20:42:44 +00:00
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2020-01-09 07:59:21 -08:00
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2026-03-10 12:44:51 +00:00
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2013-09-23 11:33:32 -07:00
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2026-03-10 12:44:51 +00:00
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2013-09-23 11:33:32 -07:00
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net: ioctl: Use kernel memory on protocol ioctl callbacks
Most of the ioctls to net protocols operates directly on userspace
argument (arg). Usually doing get_user()/put_user() directly in the
ioctl callback. This is not flexible, because it is hard to reuse these
functions without passing userspace buffers.
Change the "struct proto" ioctls to avoid touching userspace memory and
operate on kernel buffers, i.e., all protocol's ioctl callbacks is
adapted to operate on a kernel memory other than on userspace (so, no
more {put,get}_user() and friends being called in the ioctl callback).
This changes the "struct proto" ioctl format in the following way:
int (*ioctl)(struct sock *sk, int cmd,
- unsigned long arg);
+ int *karg);
(Important to say that this patch does not touch the "struct proto_ops"
protocols)
So, the "karg" argument, which is passed to the ioctl callback, is a
pointer allocated to kernel space memory (inside a function wrapper).
This buffer (karg) may contain input argument (copied from userspace in
a prep function) and it might return a value/buffer, which is copied
back to userspace if necessary. There is not one-size-fits-all format
(that is I am using 'may' above), but basically, there are three type of
ioctls:
1) Do not read from userspace, returns a result to userspace
2) Read an input parameter from userspace, and does not return anything
to userspace
3) Read an input from userspace, and return a buffer to userspace.
The default case (1) (where no input parameter is given, and an "int" is
returned to userspace) encompasses more than 90% of the cases, but there
are two other exceptions. Here is a list of exceptions:
* Protocol RAW:
* cmd = SIOCGETVIFCNT:
* input and output = struct sioc_vif_req
* cmd = SIOCGETSGCNT
* input and output = struct sioc_sg_req
* Explanation: for the SIOCGETVIFCNT case, userspace passes the input
argument, which is struct sioc_vif_req. Then the callback populates
the struct, which is copied back to userspace.
* Protocol RAW6:
* cmd = SIOCGETMIFCNT_IN6
* input and output = struct sioc_mif_req6
* cmd = SIOCGETSGCNT_IN6
* input and output = struct sioc_sg_req6
* Protocol PHONET:
* cmd == SIOCPNADDRESOURCE | SIOCPNDELRESOURCE
* input int (4 bytes)
* Nothing is copied back to userspace.
For the exception cases, functions sock_sk_ioctl_inout() will
copy the userspace input, and copy it back to kernel space.
The wrapper that prepare the buffer and put the buffer back to user is
sk_ioctl(), so, instead of calling sk->sk_prot->ioctl(), the callee now
calls sk_ioctl(), which will handle all cases.
Signed-off-by: Breno Leitao <leitao@debian.org>
Reviewed-by: Willem de Bruijn <willemb@google.com>
Reviewed-by: David Ahern <dsahern@kernel.org>
Reviewed-by: Kuniyuki Iwashima <kuniyu@amazon.com>
Link: https://lore.kernel.org/r/20230609152800.830401-1-leitao@debian.org
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2023-06-09 08:27:42 -07:00
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2024-02-26 11:22:25 +08:00
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2018-05-29 23:27:31 +08:00
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2025-10-28 12:58:01 +01:00
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2013-09-23 11:33:32 -07:00
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2024-03-29 15:32:03 +00:00
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2026-01-30 20:24:26 +01:00
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2013-09-23 11:33:32 -07:00
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2023-06-09 20:42:46 +00:00
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2021-10-25 15:13:40 -07:00
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2007-11-06 23:30:13 -08:00
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2023-10-01 11:12:38 -04:00
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2005-04-16 15:20:36 -07:00
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2005-08-09 20:10:42 -07:00
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2005-07-05 15:17:45 -07:00
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2005-08-09 20:10:42 -07:00
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2023-10-01 11:12:38 -04:00
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2005-08-09 20:10:42 -07:00
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2005-07-05 15:17:45 -07:00
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2005-08-09 20:10:42 -07:00
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2005-07-05 15:17:45 -07:00
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2005-08-09 20:10:42 -07:00
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2005-04-16 15:20:36 -07:00
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2025-03-05 23:38:47 +01:00
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2007-05-27 02:04:16 -07:00
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2009-11-03 03:26:03 +00:00
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2005-04-16 15:20:36 -07:00
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2025-03-05 23:38:51 +01:00
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2005-04-16 15:20:36 -07:00
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2013-09-23 11:33:32 -07:00
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2024-04-07 09:33:22 +00:00
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tcp: add TCP_RFC7323_TW_PAWS drop reason
Devices in the networking path, such as firewalls, NATs, or routers, which
can perform SNAT or DNAT, use addresses from their own limited address
pools to masquerade the source address during forwarding, causing PAWS
verification to fail more easily.
Currently, packet loss statistics for PAWS can only be viewed through MIB,
which is a global metric and cannot be precisely obtained through tracing
to get the specific 4-tuple of the dropped packet. In the past, we had to
use kprobe ret to retrieve relevant skb information from
tcp_timewait_state_process().
We add a drop_reason pointer, similar to what previous commit does:
commit e34100c2ecbb ("tcp: add a drop_reason pointer to tcp_check_req()")
This commit addresses the PAWSESTABREJECTED case and also sets the
corresponding drop reason.
We use 'pwru' to test.
Before this commit:
''''
./pwru 'port 9999'
2025/04/07 13:40:19 Listening for events..
TUPLE FUNC
172.31.75.115:12345->172.31.75.114:9999(tcp) sk_skb_reason_drop(SKB_DROP_REASON_NOT_SPECIFIED)
'''
After this commit:
'''
./pwru 'port 9999'
2025/04/07 13:51:34 Listening for events..
TUPLE FUNC
172.31.75.115:12345->172.31.75.114:9999(tcp) sk_skb_reason_drop(SKB_DROP_REASON_TCP_RFC7323_TW_PAWS)
'''
Suggested-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Jiayuan Chen <jiayuan.chen@linux.dev>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250409112614.16153-2-jiayuan.chen@linux.dev
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2025-04-09 19:26:04 +08:00
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2013-09-23 11:33:32 -07:00
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2018-02-13 06:14:12 -08:00
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2025-03-01 20:14:19 +00:00
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2024-02-26 11:22:26 +08:00
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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>
2014-08-04 19:12:29 -04:00
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2020-10-30 18:34:12 -07:00
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2013-09-23 11:33:32 -07:00
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2025-10-13 14:59:26 +00:00
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2013-09-23 11:33:32 -07:00
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2017-03-15 16:30:45 -04:00
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2020-11-16 10:48:04 +01:00
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2013-09-23 11:33:32 -07:00
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2020-08-20 12:00:39 -07:00
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2018-06-28 09:43:44 -07:00
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bpf: Change bpf_getsockopt(SOL_TCP) to reuse do_tcp_getsockopt()
This patch changes bpf_getsockopt(SOL_TCP) to reuse
do_tcp_getsockopt(). It removes the duplicated code from
bpf_getsockopt(SOL_TCP).
Before this patch, there were some optnames available to
bpf_setsockopt(SOL_TCP) but missing in bpf_getsockopt(SOL_TCP).
For example, TCP_NODELAY, TCP_MAXSEG, TCP_KEEPIDLE, TCP_KEEPINTVL,
and a few more. It surprises users from time to time. This patch
automatically closes this gap without duplicating more code.
bpf_getsockopt(TCP_SAVED_SYN) does not free the saved_syn,
so it stays in sol_tcp_sockopt().
For string name value like TCP_CONGESTION, bpf expects it
is always null terminated, so sol_tcp_sockopt() decrements
optlen by one before calling do_tcp_getsockopt() and
the 'if (optlen < saved_optlen) memset(..,0,..);'
in __bpf_getsockopt() will always do a null termination.
Signed-off-by: Martin KaFai Lau <martin.lau@kernel.org>
Link: https://lore.kernel.org/r/20220902002918.2894511-1-kafai@fb.com
Signed-off-by: Alexei Starovoitov <ast@kernel.org>
2022-09-01 17:29:18 -07:00
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2013-09-23 11:33:32 -07:00
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2021-01-15 08:34:59 -08:00
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2022-08-16 23:18:19 -07:00
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2020-07-23 08:09:07 +02:00
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2025-02-06 09:46:05 +00:00
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2013-09-23 11:33:32 -07:00
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2015-03-22 10:22:19 -07:00
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net: remove noblock parameter from recvmsg() entities
The internal recvmsg() functions have two parameters 'flags' and 'noblock'
that were merged inside skb_recv_datagram(). As a follow up patch to commit
f4b41f062c42 ("net: remove noblock parameter from skb_recv_datagram()")
this patch removes the separate 'noblock' parameter for recvmsg().
Analogue to the referenced patch for skb_recv_datagram() the 'flags' and
'noblock' parameters are unnecessarily split up with e.g.
err = sk->sk_prot->recvmsg(sk, msg, size, flags & MSG_DONTWAIT,
flags & ~MSG_DONTWAIT, &addr_len);
or in
err = INDIRECT_CALL_2(sk->sk_prot->recvmsg, tcp_recvmsg, udp_recvmsg,
sk, msg, size, flags & MSG_DONTWAIT,
flags & ~MSG_DONTWAIT, &addr_len);
instead of simply using only flags all the time and check for MSG_DONTWAIT
where needed (to preserve for the formerly separated no(n)block condition).
Signed-off-by: Oliver Hartkopp <socketcan@hartkopp.net>
Link: https://lore.kernel.org/r/20220411124955.154876-1-socketcan@hartkopp.net
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2022-04-11 14:49:55 +02:00
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2026-02-27 15:11:20 +00:00
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2018-04-16 10:33:35 -07:00
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2026-04-07 17:14:38 -07:00
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2020-12-02 13:31:51 -08:00
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2026-03-04 01:27:47 +00:00
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2021-06-03 16:24:31 -07:00
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tcp: avoid extra wakeups for SO_RCVLOWAT users
SO_RCVLOWAT is properly handled in tcp_poll(), so that POLLIN is only
generated when enough bytes are available in receive queue, after
David change (commit c7004482e8dc "tcp: Respect SO_RCVLOWAT in tcp_poll().")
But TCP still calls sk->sk_data_ready() for each chunk added in receive
queue, meaning thread is awaken, and goes back to sleep shortly after.
Tested:
tcp_mmap test program, receiving 32768 MB of data with SO_RCVLOWAT set to 512KB
-> Should get ~2 wakeups (c-switches) per MB, regardless of how many
(tiny or big) packets were received.
High speed (mostly full size GRO packets)
received 32768 MB (100 % mmap'ed) in 8.03112 s, 34.2266 Gbit,
cpu usage user:0.037 sys:1.404, 43.9758 usec per MB, 65497 c-switches
received 32768 MB (99.9954 % mmap'ed) in 7.98453 s, 34.4263 Gbit,
cpu usage user:0.03 sys:1.422, 44.3115 usec per MB, 65485 c-switches
Low speed (sender is ratelimited and sends 1-MSS at a time, so GRO is not helping)
received 22474.5 MB (100 % mmap'ed) in 6015.35 s, 0.0313414 Gbit,
cpu usage user:0.05 sys:1.586, 72.7952 usec per MB, 44950 c-switches
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2018-04-16 10:33:37 -07:00
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2019-01-17 18:03:14 +08:00
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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>
2018-04-16 10:33:38 -07:00
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2019-01-17 18:03:14 +08:00
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2017-06-07 10:34:36 -07:00
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2013-09-23 11:33:32 -07:00
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2008-04-17 12:29:53 +09:00
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2019-07-29 09:59:14 -07:00
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2022-06-15 16:48:44 +03:00
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2019-07-29 09:59:14 -07:00
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2005-04-16 15:20:36 -07:00
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2014-08-14 12:40:05 -04:00
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2016-02-02 19:31:12 -08:00
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2020-05-27 17:34:58 -07:00
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2013-09-23 11:33:32 -07:00
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2015-09-29 07:42:47 -07:00
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2013-09-23 11:33:32 -07:00
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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>
2015-01-05 23:57:48 +01:00
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2015-09-29 07:42:48 -07:00
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2013-09-23 11:33:32 -07:00
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2015-10-22 08:20:46 -07:00
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2026-02-17 16:12:05 +00:00
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2013-09-23 11:33:32 -07:00
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2025-11-03 16:26:13 -08:00
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2013-09-23 11:33:32 -07:00
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2016-04-13 22:05:39 -07:00
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2026-01-31 23:25:08 +01:00
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2016-04-13 22:05:39 -07:00
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2015-09-25 07:39:19 -07:00
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2013-09-23 11:33:32 -07:00
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2015-10-02 11:43:35 -07:00
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2020-08-20 12:00:52 -07:00
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2013-09-23 11:33:32 -07:00
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2005-04-16 15:20:36 -07:00
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2012-04-19 03:40:01 +00:00
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2012-05-10 01:49:41 +00:00
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2012-08-09 14:11:00 +00:00
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2005-04-16 15:20:36 -07:00
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2015-06-04 18:30:43 -07:00
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2023-11-28 18:29:21 -08:00
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2023-11-28 18:29:20 -08:00
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2014-10-15 14:33:22 -07:00
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2020-07-30 21:25:55 +02:00
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2023-11-28 18:29:24 -08:00
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2024-01-18 13:17:51 -08:00
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2011-09-18 21:02:55 -04:00
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2013-09-20 22:32:55 +02:00
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2014-03-19 21:02:21 -07:00
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2013-09-20 22:32:55 +02:00
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2014-03-19 21:02:21 -07:00
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2013-09-20 22:32:55 +02:00
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2015-05-14 14:26:56 -07:00
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2015-09-29 07:42:49 -07:00
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2015-05-14 14:26:56 -07:00
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2015-09-29 07:42:49 -07:00
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2015-05-14 14:26:56 -07:00
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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 01:01:21 -07:00
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2018-07-11 12:16:12 +02:00
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2015-05-14 14:26:56 -07:00
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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 01:01:21 -07:00
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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 12:38:36 +01:00
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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 01:01:21 -07:00
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2019-12-06 12:38:49 +01:00
|
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|
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 12:38:36 +01:00
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2023-03-17 15:55:39 +00:00
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2015-05-14 14:26:56 -07:00
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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 01:01:21 -07:00
|
|
|
|
2018-07-11 12:16:12 +02:00
|
|
|
|
2015-05-14 14:26:56 -07:00
|
|
|
|
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 01:01:21 -07:00
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|
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 12:38:43 +01:00
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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 01:01:21 -07:00
|
|
|
|
|
|
|
|
|
|
|
|
|
|
2019-12-06 12:38:49 +01:00
|
|
|
|
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 12:38:43 +01:00
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2015-05-14 14:26:56 -07:00
|
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2013-09-20 22:32:55 +02:00
|
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2014-03-19 21:02:21 -07:00
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2015-05-14 14:26:56 -07:00
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2014-03-19 21:02:21 -07:00
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2013-09-20 22:32:55 +02:00
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2024-01-15 12:55:09 -08:00
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2013-09-23 11:33:32 -07:00
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2015-09-29 07:42:49 -07:00
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2019-11-07 11:51:18 -08:00
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2017-06-07 10:34:37 -07:00
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2023-11-28 18:29:24 -08:00
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2008-04-10 03:12:40 -07:00
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2024-01-15 12:55:12 -08:00
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2008-02-07 21:49:26 -08:00
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2023-11-28 18:29:20 -08:00
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2013-09-23 11:33:32 -07:00
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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>
2014-11-03 17:35:02 +01:00
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2013-09-23 11:33:32 -07:00
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2015-09-29 07:42:49 -07:00
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2011-09-18 21:02:55 -04:00
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2005-04-16 15:20:36 -07:00
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2021-09-22 19:26:40 +02:00
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2013-09-23 11:33:32 -07:00
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tcp-tso: do not split TSO packets at retransmit time
Linux TCP stack painfully segments all TSO/GSO packets before retransmits.
This was fine back in the days when TSO/GSO were emerging, with their
bugs, but we believe the dark age is over.
Keeping big packets in write queues, but also in stack traversal
has a lot of benefits.
- Less memory overhead, because write queues have less skbs
- Less cpu overhead at ACK processing.
- Better SACK processing, as lot of studies mentioned how
awful linux was at this ;)
- Less cpu overhead to send the rtx packets
(IP stack traversal, netfilter traversal, drivers...)
- Better latencies in presence of losses.
- Smaller spikes in fq like packet schedulers, as retransmits
are not constrained by TCP Small Queues.
1 % packet losses are common today, and at 100Gbit speeds, this
translates to ~80,000 losses per second.
Losses are often correlated, and we see many retransmit events
leading to 1-MSS train of packets, at the time hosts are already
under stress.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-04-21 10:55:23 -07:00
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2013-09-23 11:33:32 -07:00
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2017-01-12 22:11:33 -08:00
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2013-09-23 11:33:32 -07:00
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2017-10-05 22:21:27 -07:00
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2005-04-16 15:20:36 -07:00
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2013-09-23 11:33:32 -07:00
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2015-05-06 14:26:25 -07:00
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2013-09-23 11:33:32 -07:00
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2024-04-25 11:13:36 +08:00
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2013-09-23 11:33:32 -07:00
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2025-03-05 23:38:52 +01:00
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2013-09-23 11:33:32 -07:00
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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>
2017-11-17 21:06:14 -05:00
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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>
2016-04-19 22:39:29 -07:00
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2005-04-16 15:20:36 -07:00
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2005-07-05 15:18:51 -07:00
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2013-09-23 11:33:32 -07:00
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2015-09-18 11:36:14 -07:00
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2024-05-28 12:52:50 +00:00
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2020-12-10 14:25:03 -08:00
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2016-02-06 11:16:28 -08:00
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2026-02-03 05:09:32 +00:00
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2023-05-31 16:01:50 +08:00
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2005-07-05 15:18:51 -07:00
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tcp: drop secpath at the same time as we currently drop dst
Xiumei reported hitting the WARN in xfrm6_tunnel_net_exit while
running tests that boil down to:
- create a pair of netns
- run a basic TCP test over ipcomp6
- delete the pair of netns
The xfrm_state found on spi_byaddr was not deleted at the time we
delete the netns, because we still have a reference on it. This
lingering reference comes from a secpath (which holds a ref on the
xfrm_state), which is still attached to an skb. This skb is not
leaked, it ends up on sk_receive_queue and then gets defer-free'd by
skb_attempt_defer_free.
The problem happens when we defer freeing an skb (push it on one CPU's
defer_list), and don't flush that list before the netns is deleted. In
that case, we still have a reference on the xfrm_state that we don't
expect at this point.
We already drop the skb's dst in the TCP receive path when it's no
longer needed, so let's also drop the secpath. At this point,
tcp_filter has already called into the LSM hooks that may require the
secpath, so it should not be needed anymore. However, in some of those
places, the MPTCP extension has just been attached to the skb, so we
cannot simply drop all extensions.
Fixes: 68822bdf76f1 ("net: generalize skb freeing deferral to per-cpu lists")
Reported-by: Xiumei Mu <xmu@redhat.com>
Signed-off-by: Sabrina Dubroca <sd@queasysnail.net>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/5055ba8f8f72bdcb602faa299faca73c280b7735.1739743613.git.sd@queasysnail.net
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-02-17 11:23:35 +01:00
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2005-04-16 15:20:36 -07:00
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2013-09-23 11:33:32 -07:00
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2005-08-09 20:10:42 -07:00
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2018-05-10 14:59:43 -07:00
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2018-05-17 14:47:24 -07:00
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2018-05-10 14:59:43 -07:00
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tcp: add SACK compression
When TCP receives an out-of-order packet, it immediately sends
a SACK packet, generating network load but also forcing the
receiver to send 1-MSS pathological packets, increasing its
RTX queue length/depth, and thus processing time.
Wifi networks suffer from this aggressive behavior, but generally
speaking, all these SACK packets add fuel to the fire when networks
are under congestion.
This patch adds a high resolution timer and tp->compressed_ack counter.
Instead of sending a SACK, we program this timer with a small delay,
based on RTT and capped to 1 ms :
delay = min ( 5 % of RTT, 1 ms)
If subsequent SACKs need to be sent while the timer has not yet
expired, we simply increment tp->compressed_ack.
When timer expires, a SACK is sent with the latest information.
Whenever an ACK is sent (if data is sent, or if in-order
data is received) timer is canceled.
Note that tcp_sack_new_ofo_skb() is able to force a SACK to be sent
if the sack blocks need to be shuffled, even if the timer has not
expired.
A new SNMP counter is added in the following patch.
Two other patches add sysctls to allow changing the 1,000,000 and 44
values that this commit hard-coded.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Toke Høiland-Jørgensen <toke@toke.dk>
Signed-off-by: David S. Miller <davem@davemloft.net>
2018-05-17 14:47:26 -07:00
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2005-08-09 20:10:42 -07:00
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2005-04-16 15:20:36 -07:00
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2013-09-23 11:33:32 -07:00
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2021-01-22 11:13:06 -08:00
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2009-03-14 14:23:05 +00:00
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2010-09-15 10:27:52 -07:00
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2016-06-28 23:06:48 +00:00
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2010-09-15 10:27:52 -07:00
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2009-03-14 14:23:05 +00:00
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2005-04-16 15:20:36 -07:00
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[INET_DIAG]: Move the tcp_diag interface to the proper place
With this the previous setup is back, i.e. tcp_diag can be built as a module,
as dccp_diag and both share the infrastructure available in inet_diag.
If one selects CONFIG_INET_DIAG as module CONFIG_INET_TCP_DIAG will also be
built as a module, as will CONFIG_INET_DCCP_DIAG, if CONFIG_IP_DCCP was
selected static or as a module, if CONFIG_INET_DIAG is y, being statically
linked CONFIG_INET_TCP_DIAG will follow suit and CONFIG_INET_DCCP_DIAG will be
built in the same manner as CONFIG_IP_DCCP.
Now to aim at UDP, converting it to use inet_hashinfo, so that we can use
iproute2 for UDP sockets as well.
Ah, just to show an example of this new infrastructure working for DCCP :-)
[root@qemu ~]# ./ss -dane
State Recv-Q Send-Q Local Address:Port Peer Address:Port
LISTEN 0 0 *:5001 *:* ino:942 sk:cfd503a0
ESTAB 0 0 127.0.0.1:5001 127.0.0.1:32770 ino:943 sk:cfd50a60
ESTAB 0 0 127.0.0.1:32770 127.0.0.1:5001 ino:947 sk:cfd50700
TIME-WAIT 0 0 127.0.0.1:32769 127.0.0.1:5001 timer:(timewait,3.430ms,0) ino:0 sk:cf209620
Signed-off-by: Arnaldo Carvalho de Melo <acme@mandriva.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2005-08-12 12:59:17 -03:00
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2015-04-28 15:28:17 -07:00
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2026-01-21 09:59:23 +00:00
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2005-04-16 15:20:36 -07:00
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2013-09-23 11:33:32 -07:00
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bpf: Fix wrong copied_seq calculation
'sk->copied_seq' was updated in the tcp_eat_skb() function when the action
of a BPF program was SK_REDIRECT. For other actions, like SK_PASS, the
update logic for 'sk->copied_seq' was moved to tcp_bpf_recvmsg_parser()
to ensure the accuracy of the 'fionread' feature.
It works for a single stream_verdict scenario, as it also modified
sk_data_ready->sk_psock_verdict_data_ready->tcp_read_skb
to remove updating 'sk->copied_seq'.
However, for programs where both stream_parser and stream_verdict are
active (strparser purpose), tcp_read_sock() was used instead of
tcp_read_skb() (sk_data_ready->strp_data_ready->tcp_read_sock).
tcp_read_sock() now still updates 'sk->copied_seq', leading to duplicate
updates.
In summary, for strparser + SK_PASS, copied_seq is redundantly calculated
in both tcp_read_sock() and tcp_bpf_recvmsg_parser().
The issue causes incorrect copied_seq calculations, which prevent
correct data reads from the recv() interface in user-land.
We do not want to add new proto_ops to implement a new version of
tcp_read_sock, as this would introduce code complexity [1].
We could have added noack and copied_seq to desc, and then called
ops->read_sock. However, unfortunately, other modules didn’t fully
initialize desc to zero. So, for now, we are directly calling
tcp_read_sock_noack() in tcp_bpf.c.
[1]: https://lore.kernel.org/bpf/20241218053408.437295-1-mrpre@163.com
Fixes: e5c6de5fa025 ("bpf, sockmap: Incorrectly handling copied_seq")
Suggested-by: Jakub Sitnicki <jakub@cloudflare.com>
Signed-off-by: Jiayuan Chen <mrpre@163.com>
Signed-off-by: Martin KaFai Lau <martin.lau@kernel.org>
Reviewed-by: Jakub Sitnicki <jakub@cloudflare.com>
Acked-by: John Fastabend <john.fastabend@gmail.com>
Link: https://patch.msgid.link/20250122100917.49845-3-mrpre@163.com
2025-01-22 18:09:14 +08:00
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2022-06-15 09:20:12 -07:00
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2022-07-22 16:50:31 -07:00
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2005-04-16 15:20:36 -07:00
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2013-09-23 11:33:32 -07:00
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2005-04-16 15:20:36 -07:00
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2013-09-23 11:33:32 -07:00
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2006-03-20 17:53:41 -08:00
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2025-02-07 15:28:29 +00:00
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2024-03-29 14:49:31 +00:00
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Revert Backoff [v3]: Revert RTO on ICMP destination unreachable
Here, an ICMP host/network unreachable message, whose payload fits to
TCP's SND.UNA, is taken as an indication that the RTO retransmission has
not been lost due to congestion, but because of a route failure
somewhere along the path.
With true congestion, a router won't trigger such a message and the
patched TCP will operate as standard TCP.
This patch reverts one RTO backoff, if an ICMP host/network unreachable
message, whose payload fits to TCP's SND.UNA, arrives.
Based on the new RTO, the retransmission timer is reset to reflect the
remaining time, or - if the revert clocked out the timer - a retransmission
is sent out immediately.
Backoffs are only reverted, if TCP is in RTO loss recovery, i.e. if
there have been retransmissions and reversible backoffs, already.
Changes from v2:
1) Renaming of skb in tcp_v4_err() moved to another patch.
2) Reintroduced tcp_bound_rto() and __tcp_set_rto().
3) Fixed code comments.
Signed-off-by: Damian Lukowski <damian@tvk.rwth-aachen.de>
Acked-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
2009-08-26 00:16:31 +00:00
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2025-02-07 15:28:29 +00:00
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Revert Backoff [v3]: Revert RTO on ICMP destination unreachable
Here, an ICMP host/network unreachable message, whose payload fits to
TCP's SND.UNA, is taken as an indication that the RTO retransmission has
not been lost due to congestion, but because of a route failure
somewhere along the path.
With true congestion, a router won't trigger such a message and the
patched TCP will operate as standard TCP.
This patch reverts one RTO backoff, if an ICMP host/network unreachable
message, whose payload fits to TCP's SND.UNA, arrives.
Based on the new RTO, the retransmission timer is reset to reflect the
remaining time, or - if the revert clocked out the timer - a retransmission
is sent out immediately.
Backoffs are only reverted, if TCP is in RTO loss recovery, i.e. if
there have been retransmissions and reversible backoffs, already.
Changes from v2:
1) Renaming of skb in tcp_v4_err() moved to another patch.
2) Reintroduced tcp_bound_rto() and __tcp_set_rto().
3) Fixed code comments.
Signed-off-by: Damian Lukowski <damian@tvk.rwth-aachen.de>
Acked-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
2009-08-26 00:16:31 +00:00
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2014-02-26 14:02:48 -08:00
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Revert Backoff [v3]: Revert RTO on ICMP destination unreachable
Here, an ICMP host/network unreachable message, whose payload fits to
TCP's SND.UNA, is taken as an indication that the RTO retransmission has
not been lost due to congestion, but because of a route failure
somewhere along the path.
With true congestion, a router won't trigger such a message and the
patched TCP will operate as standard TCP.
This patch reverts one RTO backoff, if an ICMP host/network unreachable
message, whose payload fits to TCP's SND.UNA, arrives.
Based on the new RTO, the retransmission timer is reset to reflect the
remaining time, or - if the revert clocked out the timer - a retransmission
is sent out immediately.
Backoffs are only reverted, if TCP is in RTO loss recovery, i.e. if
there have been retransmissions and reversible backoffs, already.
Changes from v2:
1) Renaming of skb in tcp_v4_err() moved to another patch.
2) Reintroduced tcp_bound_rto() and __tcp_set_rto().
3) Fixed code comments.
Signed-off-by: Damian Lukowski <damian@tvk.rwth-aachen.de>
Acked-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: David S. Miller <davem@davemloft.net>
2009-08-26 00:16:31 +00:00
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2025-11-06 00:32:43 +00:00
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2025-11-06 00:32:44 +00:00
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2025-11-06 00:32:43 +00:00
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tcp: derive delack_max from rto_min
While BPF allows to set icsk->->icsk_delack_max
and/or icsk->icsk_rto_min, we have an ip route
attribute (RTAX_RTO_MIN) to be able to tune rto_min,
but nothing to consequently adjust max delayed ack,
which vary from 40ms to 200 ms (TCP_DELACK_{MIN|MAX}).
This makes RTAX_RTO_MIN of almost no practical use,
unless customers are in big trouble.
Modern days datacenter communications want to set
rto_min to ~5 ms, and the max delayed ack one jiffie
smaller to avoid spurious retransmits.
After this patch, an "rto_min 5" route attribute will
effectively lower max delayed ack timers to 4 ms.
Note in the following ss output, "rto:6 ... ato:4"
$ ss -temoi dst XXXXXX
State Recv-Q Send-Q Local Address:Port Peer Address:Port Process
ESTAB 0 0 [2002:a05:6608:295::]:52950 [2002:a05:6608:297::]:41597
ino:255134 sk:1001 <->
skmem:(r0,rb1707063,t872,tb262144,f0,w0,o0,bl0,d0) ts sack
cubic wscale:8,8 rto:6 rtt:0.02/0.002 ato:4 mss:4096 pmtu:4500
rcvmss:536 advmss:4096 cwnd:10 bytes_sent:54823160 bytes_acked:54823121
bytes_received:54823120 segs_out:1370582 segs_in:1370580
data_segs_out:1370579 data_segs_in:1370578 send 16.4Gbps
pacing_rate 32.6Gbps delivery_rate 1.72Gbps delivered:1370579
busy:26920ms unacked:1 rcv_rtt:34.615 rcv_space:65920
rcv_ssthresh:65535 minrtt:0.015 snd_wnd:65536
While we could argue this patch fixes a bug with RTAX_RTO_MIN,
I do not add a Fixes: tag, so that we can soak it a bit before
asking backports to stable branches.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2023-09-20 17:29:43 +00:00
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2009-05-04 11:11:01 -07:00
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2023-09-20 17:29:42 +00:00
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2009-05-04 11:11:01 -07:00
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2011-10-21 05:22:42 -04:00
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2025-03-17 20:03:13 +08:00
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2009-05-04 11:11:01 -07:00
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2023-09-20 17:29:42 +00:00
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2014-02-26 14:02:48 -08:00
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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>
2015-01-05 23:57:48 +01:00
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2015-10-16 21:57:42 -07:00
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2016-09-19 23:39:10 -04:00
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2015-10-16 21:57:42 -07:00
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2005-04-16 15:20:36 -07:00
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2006-01-03 16:03:49 -08:00
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2005-04-16 15:20:36 -07:00
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tcp: implement RFC 7323 window retraction receiver requirements
By default, the Linux TCP implementation does not shrink the
advertised window (RFC 7323 calls this "window retraction") with the
following exceptions:
- When an incoming segment cannot be added due to the receive buffer
running out of memory. Since commit 8c670bdfa58e ("tcp: correct
handling of extreme memory squeeze") a zero window will be
advertised in this case. It turns out that reaching the required
memory pressure is easy when window scaling is in use. In the
simplest case, sending a sufficient number of segments smaller than
the scale factor to a receiver that does not read data is enough.
- Commit b650d953cd39 ("tcp: enforce receive buffer memory limits by
allowing the tcp window to shrink") addressed the "eating memory"
problem by introducing a sysctl knob that allows shrinking the
window before running out of memory.
However, RFC 7323 does not only state that shrinking the window is
necessary in some cases, it also formulates requirements for TCP
implementations when doing so (Section 2.4).
This commit addresses the receiver-side requirements: After retracting
the window, the peer may have a snd_nxt that lies within a previously
advertised window but is now beyond the retracted window. This means
that all incoming segments (including pure ACKs) will be rejected
until the application happens to read enough data to let the peer's
snd_nxt be in window again (which may be never).
To comply with RFC 7323, the receiver MUST honor any segment that
would have been in window for any ACK sent by the receiver and, when
window scaling is in effect, SHOULD track the maximum window sequence
number it has advertised. This patch tracks that maximum window
sequence number rcv_mwnd_seq throughout the connection and uses it in
tcp_sequence() when deciding whether a segment is acceptable.
rcv_mwnd_seq is updated together with rcv_wup and rcv_wnd in
tcp_select_window(). If we count tcp_sequence() as fast path, it is
read in the fast path. Therefore, rcv_mwnd_seq is put into rcv_wnd's
cacheline group.
The logic for handling received data in tcp_data_queue() is already
sufficient and does not need to be updated.
Signed-off-by: Simon Baatz <gmbnomis@gmail.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20260309-tcp_rfc7323_retract_wnd_rfc-v3-1-4c7f96b1ec69@gmail.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2026-03-09 09:02:26 +01:00
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2005-04-16 15:20:36 -07:00
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2013-09-23 11:33:32 -07:00
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2005-04-16 15:20:36 -07:00
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2012-04-19 03:40:39 +00:00
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2017-05-16 14:00:01 -07:00
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2017-05-16 14:00:14 -07:00
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2005-04-16 15:20:36 -07:00
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2017-05-16 14:00:14 -07:00
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2018-09-28 10:28:44 -07:00
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2017-05-16 14:00:14 -07:00
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2023-10-20 12:57:39 +00:00
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2023-10-20 12:57:40 +00:00
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2023-10-20 12:57:38 +00:00
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2023-10-20 12:57:43 +00:00
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2023-10-20 12:57:47 +00:00
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2023-10-20 12:57:43 +00:00
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2026-04-29 01:08:09 +00:00
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2018-09-21 08:51:49 -07:00
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2017-05-16 14:00:14 -07:00
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2005-04-16 15:20:36 -07:00
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2018-09-21 08:51:47 -07:00
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2018-09-21 08:51:50 -07:00
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2018-09-21 08:51:47 -07:00
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2023-10-20 12:57:41 +00:00
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2023-10-20 12:57:40 +00:00
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2023-10-20 12:57:47 +00:00
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2023-10-20 12:57:40 +00:00
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2023-10-20 12:57:47 +00:00
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2023-10-20 12:57:40 +00:00
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2014-09-05 15:33:33 -07:00
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2010-06-12 14:01:43 +00:00
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2025-03-05 23:38:43 +01:00
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2025-03-05 23:38:44 +01:00
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2010-06-12 14:01:43 +00:00
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2025-03-05 23:38:44 +01:00
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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>
2015-05-19 21:04:22 +02:00
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tcp: accecn: AccECN negotiation
Accurate ECN negotiation parts based on the specification:
https://tools.ietf.org/id/draft-ietf-tcpm-accurate-ecn-28.txt
Accurate ECN is negotiated using ECE, CWR and AE flags in the
TCP header. TCP falls back into using RFC3168 ECN if one of the
ends supports only RFC3168-style ECN.
The AccECN negotiation includes reflecting IP ECN field value
seen in SYN and SYNACK back using the same bits as negotiation
to allow responding to SYN CE marks and to detect ECN field
mangling. CE marks should not occur currently because SYN=1
segments are sent with Non-ECT in IP ECN field (but proposal
exists to remove this restriction).
Reflecting SYN IP ECN field in SYNACK is relatively simple.
Reflecting SYNACK IP ECN field in the final/third ACK of
the handshake is more challenging. Linux TCP code is not well
prepared for using the final/third ACK a signalling channel
which makes things somewhat complicated here.
tcp_ecn sysctl can be used to select the highest ECN variant
(Accurate ECN, ECN, No ECN) that is attemped to be negotiated and
requested for incoming connection and outgoing connection:
TCP_ECN_IN_NOECN_OUT_NOECN, TCP_ECN_IN_ECN_OUT_ECN,
TCP_ECN_IN_ECN_OUT_NOECN, TCP_ECN_IN_ACCECN_OUT_ACCECN,
TCP_ECN_IN_ACCECN_OUT_ECN, and TCP_ECN_IN_ACCECN_OUT_NOECN.
After this patch, the size of tcp_request_sock remains unchanged
and no new holes are added. Below are the pahole outcomes before
and after this patch:
[BEFORE THIS PATCH]
struct tcp_request_sock {
[...]
u32 rcv_nxt; /* 352 4 */
u8 syn_tos; /* 356 1 */
/* size: 360, cachelines: 6, members: 16 */
}
[AFTER THIS PATCH]
struct tcp_request_sock {
[...]
u32 rcv_nxt; /* 352 4 */
u8 syn_tos; /* 356 1 */
bool accecn_ok; /* 357 1 */
u8 syn_ect_snt:2; /* 358: 0 1 */
u8 syn_ect_rcv:2; /* 358: 2 1 */
u8 accecn_fail_mode:4; /* 358: 4 1 */
/* size: 360, cachelines: 6, members: 20 */
}
After this patch, the size of tcp_sock remains unchanged and no new
holes are added. Also, 4 bits of the existing 2-byte hole are exploited.
Below are the pahole outcomes before and after this patch:
[BEFORE THIS PATCH]
struct tcp_sock {
[...]
u8 dup_ack_counter:2; /* 2761: 0 1 */
u8 tlp_retrans:1; /* 2761: 2 1 */
u8 unused:5; /* 2761: 3 1 */
u8 thin_lto:1; /* 2762: 0 1 */
u8 fastopen_connect:1; /* 2762: 1 1 */
u8 fastopen_no_cookie:1; /* 2762: 2 1 */
u8 fastopen_client_fail:2; /* 2762: 3 1 */
u8 frto:1; /* 2762: 5 1 */
/* XXX 2 bits hole, try to pack */
[...]
u8 keepalive_probes; /* 2765 1 */
/* XXX 2 bytes hole, try to pack */
[...]
/* size: 3200, cachelines: 50, members: 164 */
}
[AFTER THIS PATCH]
struct tcp_sock {
[...]
u8 dup_ack_counter:2; /* 2761: 0 1 */
u8 tlp_retrans:1; /* 2761: 2 1 */
u8 syn_ect_snt:2; /* 2761: 3 1 */
u8 syn_ect_rcv:2; /* 2761: 5 1 */
u8 thin_lto:1; /* 2761: 7 1 */
u8 fastopen_connect:1; /* 2762: 0 1 */
u8 fastopen_no_cookie:1; /* 2762: 1 1 */
u8 fastopen_client_fail:2; /* 2762: 2 1 */
u8 frto:1; /* 2762: 4 1 */
/* XXX 3 bits hole, try to pack */
[...]
u8 keepalive_probes; /* 2765 1 */
u8 accecn_fail_mode:4; /* 2766: 0 1 */
/* XXX 4 bits hole, try to pack */
/* XXX 1 byte hole, try to pack */
[...]
/* size: 3200, cachelines: 50, members: 166 */
}
Signed-off-by: Ilpo Järvinen <ij@kernel.org>
Co-developed-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Signed-off-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Co-developed-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Signed-off-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Acked-by: Paolo Abeni <pabeni@redhat.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250916082434.100722-3-chia-yu.chang@nokia-bell-labs.com
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-09-16 10:24:26 +02:00
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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>
2015-05-19 21:04:22 +02:00
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tcp: AccECN core
This change implements Accurate ECN without negotiation and
AccECN Option (that will be added by later changes). Based on
AccECN specifications:
https://tools.ietf.org/id/draft-ietf-tcpm-accurate-ecn-28.txt
Accurate ECN allows feeding back the number of CE (congestion
experienced) marks accurately to the sender in contrast to
RFC3168 ECN that can only signal one marks-seen-yes/no per RTT.
Congestion control algorithms can take advantage of the accurate
ECN information to fine-tune their congestion response to avoid
drastic rate reduction when only mild congestion is encountered.
With Accurate ECN, tp->received_ce (r.cep in AccECN spec) keeps
track of how many segments have arrived with a CE mark. Accurate
ECN uses ACE field (ECE, CWR, AE) to communicate the value back
to the sender which updates tp->delivered_ce (s.cep) based on the
feedback. This signalling channel is lossy when ACE field overflow
occurs.
Conservative strategy is selected here to deal with the ACE
overflow, however, some strategies using the AccECN option later
in the overall patchset mitigate against false overflows detected.
The ACE field values on the wire are offset by
TCP_ACCECN_CEP_INIT_OFFSET. Delivered_ce/received_ce count the
real CE marks rather than forcing all downstream users to adapt
to the wire offset.
This patch uses the first 1-byte hole and the last 4-byte hole of
the tcp_sock_write_txrx for 'received_ce_pending' and 'received_ce'.
Also, the group size of tcp_sock_write_txrx is increased from
91 + 4 to 95 + 4 due to the new u32 received_ce member. Below are
the trimmed pahole outcomes before and after this patch.
[BEFORE THIS PATCH]
struct tcp_sock {
[...]
__cacheline_group_begin__tcp_sock_write_txrx[0]; /* 2521 0 */
u8 nonagle:4; /* 2521: 0 1 */
u8 rate_app_limited:1; /* 2521: 4 1 */
/* XXX 3 bits hole, try to pack */
/* XXX 2 bytes hole, try to pack */
[...]
u32 delivered_ce; /* 2576 4 */
u32 app_limited; /* 2580 4 */
u32 rcv_wnd; /* 2684 4 */
struct tcp_options_received rx_opt; /* 2688 24 */
__cacheline_group_end__tcp_sock_write_txrx[0]; /* 2612 0 */
/* XXX 4 bytes hole, try to pack */
[...]
/* size: 3200, cachelines: 50, members: 161 */
}
[AFTER THIS PATCH]
struct tcp_sock {
[...]
__cacheline_group_begin__tcp_sock_write_txrx[0]; /* 2521 0 */
u8 nonagle:4; /* 2521: 0 1 */
u8 rate_app_limited:1; /* 2521: 4 1 */
/* XXX 3 bits hole, try to pack */
/* Force alignment to the next boundary: */
u8 :0;
u8 received_ce_pending:4;/* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
/* XXX 1 byte hole, try to pack */
[...]
u32 delivered_ce; /* 2576 4 */
u32 received_ce; /* 2580 4 */
u32 app_limited; /* 2584 4 */
u32 rcv_wnd; /* 2588 4 */
struct tcp_options_received rx_opt; /* 2592 24 */
__cacheline_group_end__tcp_sock_write_txrx[0]; /* 2616 0 */
[...]
/* size: 3200, cachelines: 50, members: 164 */
}
Signed-off-by: Ilpo Järvinen <ij@kernel.org>
Co-developed-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Signed-off-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Co-developed-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Signed-off-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250916082434.100722-2-chia-yu.chang@nokia-bell-labs.com
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-09-16 10:24:25 +02:00
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tcp: accecn: AccECN option
The Accurate ECN allows echoing back the sum of bytes for
each IP ECN field value in the received packets using
AccECN option. This change implements AccECN option tx & rx
side processing without option send control related features
that are added by a later change.
Based on specification:
https://tools.ietf.org/id/draft-ietf-tcpm-accurate-ecn-28.txt
(Some features of the spec will be added in the later changes
rather than in this one).
A full-length AccECN option is always attempted but if it does
not fit, the minimum length is selected based on the counters
that have changed since the last update. The AccECN option
(with 24-bit fields) often ends in odd sizes so the option
write code tries to take advantage of some nop used to pad
the other TCP options.
The delivered_ecn_bytes pairs with received_ecn_bytes similar
to how delivered_ce pairs with received_ce. In contrast to
ACE field, however, the option is not always available to update
delivered_ecn_bytes. For ACK w/o AccECN option, the delivered
bytes calculated based on the cumulative ACK+SACK information
are assigned to one of the counters using an estimation
heuristic to select the most likely ECN byte counter. Any
estimation error is corrected when the next AccECN option
arrives. It may occur that the heuristic gets too confused
when there are enough different byte counter deltas between
ACKs with the AccECN option in which case the heuristic just
gives up on updating the counters for a while.
tcp_ecn_option sysctl can be used to select option sending
mode for AccECN: TCP_ECN_OPTION_DISABLED, TCP_ECN_OPTION_MINIMUM,
and TCP_ECN_OPTION_FULL.
This patch increases the size of tcp_info struct, as there is
no existing holes for new u32 variables. Below are the pahole
outcomes before and after this patch:
[BEFORE THIS PATCH]
struct tcp_info {
[...]
__u32 tcpi_total_rto_time; /* 244 4 */
/* size: 248, cachelines: 4, members: 61 */
}
[AFTER THIS PATCH]
struct tcp_info {
[...]
__u32 tcpi_total_rto_time; /* 244 4 */
__u32 tcpi_received_ce; /* 248 4 */
__u32 tcpi_delivered_e1_bytes; /* 252 4 */
__u32 tcpi_delivered_e0_bytes; /* 256 4 */
__u32 tcpi_delivered_ce_bytes; /* 260 4 */
__u32 tcpi_received_e1_bytes; /* 264 4 */
__u32 tcpi_received_e0_bytes; /* 268 4 */
__u32 tcpi_received_ce_bytes; /* 272 4 */
/* size: 280, cachelines: 5, members: 68 */
}
This patch uses the existing 1-byte holes in the tcp_sock_write_txrx
group for new u8 members, but adds a 4-byte hole in tcp_sock_write_rx
group after the new u32 delivered_ecn_bytes[3] member. Therefore, the
group size of tcp_sock_write_rx is increased from 96 to 112. Below
are the pahole outcomes before and after this patch:
[BEFORE THIS PATCH]
struct tcp_sock {
[...]
u8 received_ce_pending:4; /* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
/* XXX 1 byte hole, try to pack */
[...]
u32 rcv_rtt_last_tsecr; /* 2668 4 */
[...]
__cacheline_group_end__tcp_sock_write_rx[0]; /* 2728 0 */
[...]
/* size: 3200, cachelines: 50, members: 167 */
}
[AFTER THIS PATCH]
struct tcp_sock {
[...]
u8 received_ce_pending:4;/* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
u8 accecn_minlen:2; /* 2523: 0 1 */
u8 est_ecnfield:2; /* 2523: 2 1 */
u8 unused3:4; /* 2523: 4 1 */
[...]
u32 rcv_rtt_last_tsecr; /* 2668 4 */
u32 delivered_ecn_bytes[3];/* 2672 12 */
/* XXX 4 bytes hole, try to pack */
[...]
__cacheline_group_end__tcp_sock_write_rx[0]; /* 2744 0 */
[...]
/* size: 3200, cachelines: 50, members: 171 */
}
Signed-off-by: Ilpo Järvinen <ij@kernel.org>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Co-developed-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Signed-off-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250916082434.100722-7-chia-yu.chang@nokia-bell-labs.com
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-09-16 10:24:30 +02:00
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tcp: AccECN core
This change implements Accurate ECN without negotiation and
AccECN Option (that will be added by later changes). Based on
AccECN specifications:
https://tools.ietf.org/id/draft-ietf-tcpm-accurate-ecn-28.txt
Accurate ECN allows feeding back the number of CE (congestion
experienced) marks accurately to the sender in contrast to
RFC3168 ECN that can only signal one marks-seen-yes/no per RTT.
Congestion control algorithms can take advantage of the accurate
ECN information to fine-tune their congestion response to avoid
drastic rate reduction when only mild congestion is encountered.
With Accurate ECN, tp->received_ce (r.cep in AccECN spec) keeps
track of how many segments have arrived with a CE mark. Accurate
ECN uses ACE field (ECE, CWR, AE) to communicate the value back
to the sender which updates tp->delivered_ce (s.cep) based on the
feedback. This signalling channel is lossy when ACE field overflow
occurs.
Conservative strategy is selected here to deal with the ACE
overflow, however, some strategies using the AccECN option later
in the overall patchset mitigate against false overflows detected.
The ACE field values on the wire are offset by
TCP_ACCECN_CEP_INIT_OFFSET. Delivered_ce/received_ce count the
real CE marks rather than forcing all downstream users to adapt
to the wire offset.
This patch uses the first 1-byte hole and the last 4-byte hole of
the tcp_sock_write_txrx for 'received_ce_pending' and 'received_ce'.
Also, the group size of tcp_sock_write_txrx is increased from
91 + 4 to 95 + 4 due to the new u32 received_ce member. Below are
the trimmed pahole outcomes before and after this patch.
[BEFORE THIS PATCH]
struct tcp_sock {
[...]
__cacheline_group_begin__tcp_sock_write_txrx[0]; /* 2521 0 */
u8 nonagle:4; /* 2521: 0 1 */
u8 rate_app_limited:1; /* 2521: 4 1 */
/* XXX 3 bits hole, try to pack */
/* XXX 2 bytes hole, try to pack */
[...]
u32 delivered_ce; /* 2576 4 */
u32 app_limited; /* 2580 4 */
u32 rcv_wnd; /* 2684 4 */
struct tcp_options_received rx_opt; /* 2688 24 */
__cacheline_group_end__tcp_sock_write_txrx[0]; /* 2612 0 */
/* XXX 4 bytes hole, try to pack */
[...]
/* size: 3200, cachelines: 50, members: 161 */
}
[AFTER THIS PATCH]
struct tcp_sock {
[...]
__cacheline_group_begin__tcp_sock_write_txrx[0]; /* 2521 0 */
u8 nonagle:4; /* 2521: 0 1 */
u8 rate_app_limited:1; /* 2521: 4 1 */
/* XXX 3 bits hole, try to pack */
/* Force alignment to the next boundary: */
u8 :0;
u8 received_ce_pending:4;/* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
/* XXX 1 byte hole, try to pack */
[...]
u32 delivered_ce; /* 2576 4 */
u32 received_ce; /* 2580 4 */
u32 app_limited; /* 2584 4 */
u32 rcv_wnd; /* 2588 4 */
struct tcp_options_received rx_opt; /* 2592 24 */
__cacheline_group_end__tcp_sock_write_txrx[0]; /* 2616 0 */
[...]
/* size: 3200, cachelines: 50, members: 164 */
}
Signed-off-by: Ilpo Järvinen <ij@kernel.org>
Co-developed-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Signed-off-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Co-developed-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Signed-off-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250916082434.100722-2-chia-yu.chang@nokia-bell-labs.com
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-09-16 10:24:25 +02:00
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2024-04-21 12:20:08 +08:00
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2005-11-10 17:13:47 -08:00
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2010-07-15 21:41:00 -07:00
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2005-04-16 15:20:36 -07:00
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2014-09-24 04:11:22 -07:00
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tcp: fix stale per-CPU tcp_tw_isn leak enabling ISN prediction
Blamed commit moved the TIME_WAIT-derived ISN from the skb control
block to a per-CPU variable, assuming the value would always be consumed
by tcp_conn_request() for the same packet that wrote it. That assumption
is violated by multiple drop paths between the producer
(__this_cpu_write(tcp_tw_isn, isn) in tcp_v{4,6}_rcv()) and the consumer
(tcp_conn_request()):
- min_ttl / min_hopcount check
- xfrm policy check
- tcp_inbound_hash() MD5/AO mismatch
- tcp_filter() eBPF/SO_ATTACH_FILTER drop
- th->syn && th->fin discard in tcp_rcv_state_process() TCP_LISTEN
- psp_sk_rx_policy_check() in tcp_v{4,6}_do_rcv()
- tcp_checksum_complete() in tcp_v{4,6}_do_rcv()
- tcp_v{4,6}_cookie_check() returning NULL
When a packet is dropped on any of these paths, tcp_tw_isn is left set.
The next SYN processed on the same CPU then consumes the non zero value in
tcp_conn_request(), receiving a potentially predictable ISN.
This patch moves back tcp_tw_isn to skb->cb[], getting rid of the per-cpu
variable.
Note that tcp_v{4,6}_fill_cb() do not set it.
Very litle impact on overall code size/complexity:
$ scripts/bloat-o-meter -t vmlinux.old vmlinux.new
add/remove: 0/0 grow/shrink: 2/1 up/down: 8/-15 (-7)
Function old new delta
tcp_v6_rcv 3038 3042 +4
tcp_v4_rcv 3035 3039 +4
tcp_conn_request 2938 2923 -15
Total: Before=24436060, After=24436053, chg -0.00%
Fixes: 41eecbd712b7 ("tcp: replace TCP_SKB_CB(skb)->tcp_tw_isn with a per-cpu field")
Reported-by: Chris Mason <clm@meta.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Kuniyuki Iwashima <kuniyu@google.com>
Link: https://patch.msgid.link/20260519084611.2485277-1-edumazet@google.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2026-05-19 08:46:11 +00:00
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2015-06-11 09:15:18 -07:00
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2014-09-24 04:11:22 -07:00
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tcp: fix stale per-CPU tcp_tw_isn leak enabling ISN prediction
Blamed commit moved the TIME_WAIT-derived ISN from the skb control
block to a per-CPU variable, assuming the value would always be consumed
by tcp_conn_request() for the same packet that wrote it. That assumption
is violated by multiple drop paths between the producer
(__this_cpu_write(tcp_tw_isn, isn) in tcp_v{4,6}_rcv()) and the consumer
(tcp_conn_request()):
- min_ttl / min_hopcount check
- xfrm policy check
- tcp_inbound_hash() MD5/AO mismatch
- tcp_filter() eBPF/SO_ATTACH_FILTER drop
- th->syn && th->fin discard in tcp_rcv_state_process() TCP_LISTEN
- psp_sk_rx_policy_check() in tcp_v{4,6}_do_rcv()
- tcp_checksum_complete() in tcp_v{4,6}_do_rcv()
- tcp_v{4,6}_cookie_check() returning NULL
When a packet is dropped on any of these paths, tcp_tw_isn is left set.
The next SYN processed on the same CPU then consumes the non zero value in
tcp_conn_request(), receiving a potentially predictable ISN.
This patch moves back tcp_tw_isn to skb->cb[], getting rid of the per-cpu
variable.
Note that tcp_v{4,6}_fill_cb() do not set it.
Very litle impact on overall code size/complexity:
$ scripts/bloat-o-meter -t vmlinux.old vmlinux.new
add/remove: 0/0 grow/shrink: 2/1 up/down: 8/-15 (-7)
Function old new delta
tcp_v6_rcv 3038 3042 +4
tcp_v4_rcv 3035 3039 +4
tcp_conn_request 2938 2923 -15
Total: Before=24436060, After=24436053, chg -0.00%
Fixes: 41eecbd712b7 ("tcp: replace TCP_SKB_CB(skb)->tcp_tw_isn with a per-cpu field")
Reported-by: Chris Mason <clm@meta.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Kuniyuki Iwashima <kuniyu@google.com>
Link: https://patch.msgid.link/20260519084611.2485277-1-edumazet@google.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2026-05-19 08:46:11 +00:00
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2015-06-11 09:15:18 -07:00
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2014-09-24 04:11:22 -07:00
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2025-03-05 23:38:44 +01:00
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2012-04-16 07:08:06 +00:00
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2017-11-08 13:01:26 -08:00
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2012-04-16 07:08:06 +00:00
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2025-02-20 15:29:37 +08:00
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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>
2016-04-25 14:44:48 -07:00
|
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2017-08-22 17:08:48 -04:00
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2025-02-20 15:29:37 +08:00
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2005-04-16 15:20:36 -07:00
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2014-09-27 09:50:57 -07:00
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2016-05-06 20:35:35 -07:00
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2021-09-23 21:17:07 +00:00
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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>
2016-09-19 23:39:14 -04:00
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2021-09-23 21:17:07 +00:00
|
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|
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>
2016-09-19 23:39:14 -04:00
|
|
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|
|
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|
|
|
|
|
|
2017-05-16 14:00:14 -07:00
|
|
|
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-19 23:39:14 -04:00
|
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2017-05-16 14:00:14 -07:00
|
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2016-05-06 20:35:35 -07:00
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2014-09-27 09:50:57 -07:00
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2016-05-06 20:35:35 -07:00
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2014-09-27 09:50:57 -07:00
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2016-05-06 20:35:35 -07:00
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2005-04-16 15:20:36 -07:00
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2020-06-23 15:31:13 -07:00
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2014-10-18 08:34:37 -07:00
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2014-10-17 09:17:20 -07:00
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2018-07-19 12:41:18 -07:00
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2014-10-17 09:17:20 -07:00
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2016-10-16 20:02:52 -07:00
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2016-05-10 11:19:50 -07:00
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2014-10-17 09:17:20 -07:00
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2017-08-07 08:44:21 -07:00
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2020-06-19 12:12:35 -07:00
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2020-06-23 15:31:12 -07:00
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2020-06-23 15:31:11 -07:00
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2020-06-19 12:12:35 -07:00
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2014-10-18 08:34:37 -07:00
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2014-10-17 09:17:20 -07:00
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2017-08-07 08:44:17 -07:00
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2005-04-16 15:20:36 -07:00
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2014-09-24 04:11:22 -07:00
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2005-04-16 15:20:36 -07:00
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2015-06-11 09:15:18 -07:00
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2005-04-16 15:20:36 -07:00
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2015-06-11 09:15:18 -07:00
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2005-04-16 15:20:36 -07:00
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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>
2016-04-25 14:44:48 -07:00
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2020-01-09 07:59:20 -08:00
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2024-05-30 16:36:15 -07:00
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2020-01-09 07:59:20 -08:00
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2021-11-02 22:58:44 -04:00
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2024-09-10 17:14:52 +00:00
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2020-01-09 07:59:20 -08:00
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2024-05-30 16:36:14 -07:00
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2005-06-23 12:19:55 -07:00
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2014-09-26 22:37:35 +02:00
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2014-09-26 22:37:34 +02:00
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2014-09-26 22:37:35 +02:00
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2014-09-26 22:37:34 +02:00
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2017-08-30 19:24:57 +02:00
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2005-06-23 12:19:55 -07:00
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2006-11-09 16:32:06 -08:00
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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>
2015-01-05 23:57:46 +01:00
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2014-09-26 22:37:33 +02:00
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2025-03-05 23:38:43 +01:00
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2014-09-26 22:37:33 +02:00
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2025-03-05 23:38:43 +01:00
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2026-01-31 23:25:04 +01:00
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2026-01-31 23:25:05 +01:00
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2026-01-31 23:25:04 +01:00
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2026-01-31 23:25:05 +01:00
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2007-04-23 22:26:16 -07:00
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2015-04-28 16:23:48 -07:00
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2016-05-11 10:02:13 -07:00
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2016-06-08 21:16:44 -07:00
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2016-05-11 10:02:13 -07:00
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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>
2016-09-19 23:39:14 -04:00
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2017-05-16 14:00:14 -07:00
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|
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>
2016-09-19 23:39:14 -04:00
|
|
|
|
2021-09-23 21:17:07 +00:00
|
|
|
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-19 23:39:14 -04:00
|
|
|
|
2021-09-23 21:17:07 +00:00
|
|
|
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-19 23:39:14 -04:00
|
|
|
|
2018-07-09 17:53:39 +00:00
|
|
|
|
|
|
|
|
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-19 23:39:14 -04:00
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
2022-04-20 10:34:41 +08:00
|
|
|
|
2016-09-19 23:39:15 -04:00
|
|
|
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-19 23:39:14 -04:00
|
|
|
|
2018-01-17 12:11:01 -08:00
|
|
|
|
tcp: track data delivery rate for a TCP connection
This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-19 23:39:14 -04:00
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2005-06-23 12:19:55 -07:00
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2021-04-02 11:10:37 -07:00
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2026-01-05 12:55:33 +01:00
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2021-04-02 11:10:37 -07:00
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2026-03-23 23:49:20 +00:00
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2021-04-02 11:10:37 -07:00
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2026-01-05 12:55:33 +01:00
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2021-04-02 11:10:37 -07:00
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2005-06-23 12:19:55 -07:00
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2021-04-02 11:10:37 -07:00
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2005-06-23 12:19:55 -07:00
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2005-08-10 04:03:31 -03:00
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2005-06-23 12:19:55 -07:00
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2005-08-10 04:03:31 -03:00
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2021-04-02 11:10:37 -07:00
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2005-06-23 12:19:55 -07:00
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2013-09-23 11:33:32 -07:00
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2023-03-22 20:23:59 -07:00
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2005-06-23 12:19:55 -07:00
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2014-09-26 22:37:32 +02:00
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2013-09-23 11:33:32 -07:00
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2017-11-14 08:25:49 -08:00
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2013-09-23 11:33:32 -07:00
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2019-07-18 19:28:14 -07:00
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2020-09-10 15:35:34 -04:00
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2015-01-28 20:01:35 -05:00
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2005-06-23 12:19:55 -07:00
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2013-09-23 11:33:32 -07:00
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2016-11-21 14:18:38 +01:00
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2014-05-02 21:18:05 -07:00
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2005-06-23 23:45:02 -07:00
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2005-06-23 12:19:55 -07:00
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2020-01-08 16:35:08 -08:00
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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>
2015-01-05 23:57:46 +01:00
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2024-05-31 23:46:34 +08:00
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2015-01-05 23:57:47 +01:00
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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>
2015-01-05 23:57:46 +01:00
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2015-01-05 23:57:47 +01:00
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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>
2015-01-05 23:57:46 +01:00
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2014-09-26 22:37:33 +02:00
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2026-01-31 23:25:04 +01:00
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2026-01-31 23:25:05 +01:00
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2005-08-10 04:03:31 -03:00
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2005-06-23 12:19:55 -07:00
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2005-08-10 04:03:31 -03:00
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2026-03-23 23:49:20 +00:00
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2026-03-25 15:18:54 +08:00
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2026-03-23 23:49:20 +00:00
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2005-08-10 04:03:31 -03:00
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2005-06-23 12:19:55 -07:00
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2022-04-06 09:09:56 +08:00
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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>
2016-09-19 23:39:14 -04:00
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2022-04-20 10:34:41 +08:00
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2007-08-09 15:14:46 +03:00
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2018-11-27 14:42:00 -08:00
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2007-08-09 15:14:46 +03:00
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2012-05-16 23:15:34 +00:00
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2007-08-09 15:14:46 +03:00
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2007-08-09 14:37:30 +03:00
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2005-04-16 15:20:36 -07:00
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2006-01-03 16:03:49 -08:00
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2005-04-16 15:20:36 -07:00
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2007-08-09 14:37:30 +03:00
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2005-04-16 15:20:36 -07:00
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2009-09-15 01:30:10 -07:00
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2022-04-05 16:35:38 -07:00
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2026-04-16 20:03:08 +00:00
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2022-04-05 16:35:38 -07:00
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2015-07-09 13:16:29 -07:00
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2022-04-05 16:35:38 -07:00
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2015-07-09 13:16:29 -07:00
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2009-09-15 01:30:10 -07:00
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2012-09-02 17:38:04 +00:00
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2005-04-16 15:20:36 -07:00
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2012-09-02 17:38:04 +00:00
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2005-04-16 15:20:36 -07:00
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2005-08-10 04:03:31 -03:00
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2005-04-16 15:20:36 -07:00
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2005-08-10 04:03:31 -03:00
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2011-10-21 05:22:42 -04:00
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2012-09-02 17:38:04 +00:00
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2005-04-16 15:20:36 -07:00
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2022-04-05 16:35:38 -07:00
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2005-04-16 15:20:36 -07:00
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2007-07-27 16:36:17 +03:00
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2005-04-16 15:20:36 -07:00
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2014-07-14 16:58:32 +02:00
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2013-09-23 11:33:32 -07:00
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2005-04-16 15:20:36 -07:00
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2011-11-21 17:15:14 +00:00
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2007-12-31 04:48:41 -08:00
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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>
2014-04-30 11:58:13 -07:00
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2014-05-22 10:41:08 -04:00
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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>
2014-04-30 11:58:13 -07:00
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2014-05-02 21:18:05 -07:00
|
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|
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>
2014-04-30 11:58:13 -07:00
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tcp: fix tcp_cwnd_validate() to not forget is_cwnd_limited
This commit fixes a bug in the tracking of max_packets_out and
is_cwnd_limited. This bug can cause the connection to fail to remember
that is_cwnd_limited is true, causing the connection to fail to grow
cwnd when it should, causing throughput to be lower than it should be.
The following event sequence is an example that triggers the bug:
(a) The connection is cwnd_limited, but packets_out is not at its
peak due to TSO deferral deciding not to send another skb yet.
In such cases the connection can advance max_packets_seq and set
tp->is_cwnd_limited to true and max_packets_out to a small
number.
(b) Then later in the round trip the connection is pacing-limited (not
cwnd-limited), and packets_out is larger. In such cases the
connection would raise max_packets_out to a bigger number but
(unexpectedly) flip tp->is_cwnd_limited from true to false.
This commit fixes that bug.
One straightforward fix would be to separately track (a) the next
window after max_packets_out reaches a maximum, and (b) the next
window after tp->is_cwnd_limited is set to true. But this would
require consuming an extra u32 sequence number.
Instead, to save space we track only the most important
information. Specifically, we track the strongest available signal of
the degree to which the cwnd is fully utilized:
(1) If the connection is cwnd-limited then we remember that fact for
the current window.
(2) If the connection not cwnd-limited then we track the maximum
number of outstanding packets in the current window.
In particular, note that the new logic cannot trigger the buggy
(a)/(b) sequence above because with the new logic a condition where
tp->packets_out > tp->max_packets_out can only trigger an update of
tp->is_cwnd_limited if tp->is_cwnd_limited is false.
This first showed up in a testing of a BBRv2 dev branch, but this
buggy behavior highlighted a general issue with the
tcp_cwnd_validate() logic that can cause cwnd to fail to increase at
the proper rate for any TCP congestion control, including Reno or
CUBIC.
Fixes: ca8a22634381 ("tcp: make cwnd-limited checks measurement-based, and gentler")
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Kevin(Yudong) Yang <yyd@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>
2022-09-28 16:03:31 -04:00
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2014-05-22 10:41:08 -04:00
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2015-07-09 13:16:29 -07:00
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2022-04-05 16:35:38 -07:00
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|
2014-05-22 10:41:08 -04:00
|
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|
|
tcp: fix tcp_cwnd_validate() to not forget is_cwnd_limited
This commit fixes a bug in the tracking of max_packets_out and
is_cwnd_limited. This bug can cause the connection to fail to remember
that is_cwnd_limited is true, causing the connection to fail to grow
cwnd when it should, causing throughput to be lower than it should be.
The following event sequence is an example that triggers the bug:
(a) The connection is cwnd_limited, but packets_out is not at its
peak due to TSO deferral deciding not to send another skb yet.
In such cases the connection can advance max_packets_seq and set
tp->is_cwnd_limited to true and max_packets_out to a small
number.
(b) Then later in the round trip the connection is pacing-limited (not
cwnd-limited), and packets_out is larger. In such cases the
connection would raise max_packets_out to a bigger number but
(unexpectedly) flip tp->is_cwnd_limited from true to false.
This commit fixes that bug.
One straightforward fix would be to separately track (a) the next
window after max_packets_out reaches a maximum, and (b) the next
window after tp->is_cwnd_limited is set to true. But this would
require consuming an extra u32 sequence number.
Instead, to save space we track only the most important
information. Specifically, we track the strongest available signal of
the degree to which the cwnd is fully utilized:
(1) If the connection is cwnd-limited then we remember that fact for
the current window.
(2) If the connection not cwnd-limited then we track the maximum
number of outstanding packets in the current window.
In particular, note that the new logic cannot trigger the buggy
(a)/(b) sequence above because with the new logic a condition where
tp->packets_out > tp->max_packets_out can only trigger an update of
tp->is_cwnd_limited if tp->is_cwnd_limited is false.
This first showed up in a testing of a BBRv2 dev branch, but this
buggy behavior highlighted a general issue with the
tcp_cwnd_validate() logic that can cause cwnd to fail to increase at
the proper rate for any TCP congestion control, including Reno or
CUBIC.
Fixes: ca8a22634381 ("tcp: make cwnd-limited checks measurement-based, and gentler")
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Kevin(Yudong) Yang <yyd@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>
2022-09-28 16:03:31 -04:00
|
|
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|
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>
2014-04-30 11:58:13 -07:00
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2005-11-10 16:53:30 -08:00
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2018-06-20 16:07:35 -04:00
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2020-05-04 11:27:49 -07:00
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2018-10-23 11:54:16 -07:00
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2020-05-04 11:27:49 -07:00
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2018-10-23 11:54:16 -07:00
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2020-05-04 11:27:49 -07:00
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2018-10-23 11:54:16 -07:00
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2020-05-04 11:27:49 -07:00
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2018-10-23 11:54:16 -07:00
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2025-02-07 15:28:27 +00:00
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2018-10-23 11:54:16 -07:00
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2025-02-07 15:28:27 +00:00
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2025-02-07 15:28:29 +00:00
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2018-10-23 11:54:16 -07:00
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tcp: adjust window probe timers to safer values
With the advent of small rto timers in datacenter TCP,
(ip route ... rto_min x), the following can happen :
1) Qdisc is full, transmit fails.
TCP sets a timer based on icsk_rto to retry the transmit, without
exponential backoff.
With low icsk_rto, and lot of sockets, all cpus are servicing timer
interrupts like crazy.
Intent of the code was to retry with a timer between 200 (TCP_RTO_MIN)
and 500ms (TCP_RESOURCE_PROBE_INTERVAL)
2) Receivers can send zero windows if they don't drain their receive queue.
TCP sends zero window probes, based on icsk_rto current value, with
exponential backoff.
With /proc/sys/net/ipv4/tcp_retries2 being 15 (or even smaller in
some cases), sender can abort in less than one or two minutes !
If receiver stops the sender, it obviously doesn't care of very tight
rto. Probability of dropping the ACK reopening the window is not
worth the risk.
Lets change the base timer to be at least 200ms (TCP_RTO_MIN) for these
events (but not normal RTO based retransmits)
A followup patch adds a new SNMP counter, as it would have helped a lot
diagnosing this issue.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-05-06 14:26:24 -07:00
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2018-10-23 11:54:16 -07:00
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|
|
tcp: adjust window probe timers to safer values
With the advent of small rto timers in datacenter TCP,
(ip route ... rto_min x), the following can happen :
1) Qdisc is full, transmit fails.
TCP sets a timer based on icsk_rto to retry the transmit, without
exponential backoff.
With low icsk_rto, and lot of sockets, all cpus are servicing timer
interrupts like crazy.
Intent of the code was to retry with a timer between 200 (TCP_RTO_MIN)
and 500ms (TCP_RESOURCE_PROBE_INTERVAL)
2) Receivers can send zero windows if they don't drain their receive queue.
TCP sends zero window probes, based on icsk_rto current value, with
exponential backoff.
With /proc/sys/net/ipv4/tcp_retries2 being 15 (or even smaller in
some cases), sender can abort in less than one or two minutes !
If receiver stops the sender, it obviously doesn't care of very tight
rto. Probability of dropping the ACK reopening the window is not
worth the risk.
Lets change the base timer to be at least 200ms (TCP_RTO_MIN) for these
events (but not normal RTO based retransmits)
A followup patch adds a new SNMP counter, as it would have helped a lot
diagnosing this issue.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-05-06 14:26:24 -07:00
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2020-12-08 17:19:10 +08:00
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tcp: adjust window probe timers to safer values
With the advent of small rto timers in datacenter TCP,
(ip route ... rto_min x), the following can happen :
1) Qdisc is full, transmit fails.
TCP sets a timer based on icsk_rto to retry the transmit, without
exponential backoff.
With low icsk_rto, and lot of sockets, all cpus are servicing timer
interrupts like crazy.
Intent of the code was to retry with a timer between 200 (TCP_RTO_MIN)
and 500ms (TCP_RESOURCE_PROBE_INTERVAL)
2) Receivers can send zero windows if they don't drain their receive queue.
TCP sends zero window probes, based on icsk_rto current value, with
exponential backoff.
With /proc/sys/net/ipv4/tcp_retries2 being 15 (or even smaller in
some cases), sender can abort in less than one or two minutes !
If receiver stops the sender, it obviously doesn't care of very tight
rto. Probability of dropping the ACK reopening the window is not
worth the risk.
Lets change the base timer to be at least 200ms (TCP_RTO_MIN) for these
events (but not normal RTO based retransmits)
A followup patch adds a new SNMP counter, as it would have helped a lot
diagnosing this issue.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-05-06 14:26:24 -07:00
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[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>
2007-04-20 22:18:02 -07:00
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2005-04-16 15:20:36 -07:00
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|
tcp: adjust window probe timers to safer values
With the advent of small rto timers in datacenter TCP,
(ip route ... rto_min x), the following can happen :
1) Qdisc is full, transmit fails.
TCP sets a timer based on icsk_rto to retry the transmit, without
exponential backoff.
With low icsk_rto, and lot of sockets, all cpus are servicing timer
interrupts like crazy.
Intent of the code was to retry with a timer between 200 (TCP_RTO_MIN)
and 500ms (TCP_RESOURCE_PROBE_INTERVAL)
2) Receivers can send zero windows if they don't drain their receive queue.
TCP sends zero window probes, based on icsk_rto current value, with
exponential backoff.
With /proc/sys/net/ipv4/tcp_retries2 being 15 (or even smaller in
some cases), sender can abort in less than one or two minutes !
If receiver stops the sender, it obviously doesn't care of very tight
rto. Probability of dropping the ACK reopening the window is not
worth the risk.
Lets change the base timer to be at least 200ms (TCP_RTO_MIN) for these
events (but not normal RTO based retransmits)
A followup patch adds a new SNMP counter, as it would have helped a lot
diagnosing this issue.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-05-06 14:26:24 -07:00
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|
|
2018-10-23 11:54:16 -07:00
|
|
|
|
2025-02-07 15:28:27 +00:00
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2005-04-16 15:20:36 -07:00
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2009-03-02 22:42:02 -08:00
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2005-04-16 15:20:36 -07:00
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2009-03-02 22:42:02 -08:00
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2005-04-16 15:20:36 -07:00
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2007-02-04 20:15:27 -08:00
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2005-04-16 15:20:36 -07:00
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2019-04-21 00:50:42 +09:00
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2005-04-16 15:20:36 -07:00
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2012-05-16 23:15:34 +00:00
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2005-04-16 15:20:36 -07:00
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2007-04-09 11:59:39 -07:00
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2018-11-12 16:17:00 -08:00
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2005-04-16 15:20:36 -07:00
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2026-04-09 10:11:47 +00:00
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tcp: defer skb freeing after socket lock is released
tcp recvmsg() (or rx zerocopy) spends a fair amount of time
freeing skbs after their payload has been consumed.
A typical ~64KB GRO packet has to release ~45 page
references, eventually going to page allocator
for each of them.
Currently, this freeing is performed while socket lock
is held, meaning that there is a high chance that
BH handler has to queue incoming packets to tcp socket backlog.
This can cause additional latencies, because the user
thread has to process the backlog at release_sock() time,
and while doing so, additional frames can be added
by BH handler.
This patch adds logic to defer these frees after socket
lock is released, or directly from BH handler if possible.
Being able to free these skbs from BH handler helps a lot,
because this avoids the usual alloc/free assymetry,
when BH handler and user thread do not run on same cpu or
NUMA node.
One cpu can now be fully utilized for the kernel->user copy,
and another cpu is handling BH processing and skb/page
allocs/frees (assuming RFS is not forcing use of a single CPU)
Tested:
100Gbit NIC
Max throughput for one TCP_STREAM flow, over 10 runs
MTU : 1500
Before: 55 Gbit
After: 66 Gbit
MTU : 4096+(headers)
Before: 82 Gbit
After: 95 Gbit
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2021-11-15 11:02:46 -08:00
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2026-04-09 14:56:23 +00:00
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2026-02-05 16:43:29 +00:00
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2026-04-09 14:56:24 +00:00
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2026-02-05 16:43:29 +00:00
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tcp: defer skb freeing after socket lock is released
tcp recvmsg() (or rx zerocopy) spends a fair amount of time
freeing skbs after their payload has been consumed.
A typical ~64KB GRO packet has to release ~45 page
references, eventually going to page allocator
for each of them.
Currently, this freeing is performed while socket lock
is held, meaning that there is a high chance that
BH handler has to queue incoming packets to tcp socket backlog.
This can cause additional latencies, because the user
thread has to process the backlog at release_sock() time,
and while doing so, additional frames can be added
by BH handler.
This patch adds logic to defer these frees after socket
lock is released, or directly from BH handler if possible.
Being able to free these skbs from BH handler helps a lot,
because this avoids the usual alloc/free assymetry,
when BH handler and user thread do not run on same cpu or
NUMA node.
One cpu can now be fully utilized for the kernel->user copy,
and another cpu is handling BH processing and skb/page
allocs/frees (assuming RFS is not forcing use of a single CPU)
Tested:
100Gbit NIC
Max throughput for one TCP_STREAM flow, over 10 runs
MTU : 1500
Before: 55 Gbit
After: 66 Gbit
MTU : 4096+(headers)
Before: 82 Gbit
After: 95 Gbit
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2021-11-15 11:02:46 -08:00
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2013-09-23 11:33:32 -07:00
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2015-12-16 12:30:05 +09:00
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2006-01-03 16:03:49 -08:00
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2005-04-16 15:20:36 -07:00
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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>
2015-08-21 12:30:00 -07:00
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2017-05-05 12:53:23 -07:00
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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>
2015-08-21 12:30:00 -07:00
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2022-07-18 10:26:48 -07:00
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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>
2015-08-21 12:30:00 -07:00
|
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2017-05-16 14:00:03 -07:00
|
|
|
|
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>
2015-08-21 12:30:00 -07:00
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2013-06-11 15:35:32 -07:00
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|
2005-04-16 15:20:36 -07:00
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|
|
2017-10-27 07:47:24 -07:00
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2013-09-23 11:33:32 -07:00
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2005-04-16 15:20:36 -07:00
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2023-07-20 20:47:50 +02:00
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2005-04-16 15:20:36 -07:00
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2023-07-20 20:47:50 +02:00
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2017-03-24 07:05:12 +08:00
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|
tcp: get rid of sysctl_tcp_adv_win_scale
With modern NIC drivers shifting to full page allocations per
received frame, we face the following issue:
TCP has one per-netns sysctl used to tweak how to translate
a memory use into an expected payload (RWIN), in RX path.
tcp_win_from_space() implementation is limited to few cases.
For hosts dealing with various MSS, we either under estimate
or over estimate the RWIN we send to the remote peers.
For instance with the default sysctl_tcp_adv_win_scale value,
we expect to store 50% of payload per allocated chunk of memory.
For the typical use of MTU=1500 traffic, and order-0 pages allocations
by NIC drivers, we are sending too big RWIN, leading to potential
tcp collapse operations, which are extremely expensive and source
of latency spikes.
This patch makes sysctl_tcp_adv_win_scale obsolete, and instead
uses a per socket scaling factor, so that we can precisely
adjust the RWIN based on effective skb->len/skb->truesize ratio.
This patch alone can double TCP receive performance when receivers
are too slow to drain their receive queue, or by allowing
a bigger RWIN when MSS is close to PAGE_SIZE.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Link: https://lore.kernel.org/r/20230717152917.751987-1-edumazet@google.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2023-07-17 15:29:17 +00:00
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2023-07-20 20:47:50 +02:00
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tcp: get rid of sysctl_tcp_adv_win_scale
With modern NIC drivers shifting to full page allocations per
received frame, we face the following issue:
TCP has one per-netns sysctl used to tweak how to translate
a memory use into an expected payload (RWIN), in RX path.
tcp_win_from_space() implementation is limited to few cases.
For hosts dealing with various MSS, we either under estimate
or over estimate the RWIN we send to the remote peers.
For instance with the default sysctl_tcp_adv_win_scale value,
we expect to store 50% of payload per allocated chunk of memory.
For the typical use of MTU=1500 traffic, and order-0 pages allocations
by NIC drivers, we are sending too big RWIN, leading to potential
tcp collapse operations, which are extremely expensive and source
of latency spikes.
This patch makes sysctl_tcp_adv_win_scale obsolete, and instead
uses a per socket scaling factor, so that we can precisely
adjust the RWIN based on effective skb->len/skb->truesize ratio.
This patch alone can double TCP receive performance when receivers
are too slow to drain their receive queue, or by allowing
a bigger RWIN when MSS is close to PAGE_SIZE.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Link: https://lore.kernel.org/r/20230717152917.751987-1-edumazet@google.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2023-07-17 15:29:17 +00:00
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2023-07-20 20:47:50 +02:00
|
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|
tcp: get rid of sysctl_tcp_adv_win_scale
With modern NIC drivers shifting to full page allocations per
received frame, we face the following issue:
TCP has one per-netns sysctl used to tweak how to translate
a memory use into an expected payload (RWIN), in RX path.
tcp_win_from_space() implementation is limited to few cases.
For hosts dealing with various MSS, we either under estimate
or over estimate the RWIN we send to the remote peers.
For instance with the default sysctl_tcp_adv_win_scale value,
we expect to store 50% of payload per allocated chunk of memory.
For the typical use of MTU=1500 traffic, and order-0 pages allocations
by NIC drivers, we are sending too big RWIN, leading to potential
tcp collapse operations, which are extremely expensive and source
of latency spikes.
This patch makes sysctl_tcp_adv_win_scale obsolete, and instead
uses a per socket scaling factor, so that we can precisely
adjust the RWIN based on effective skb->len/skb->truesize ratio.
This patch alone can double TCP receive performance when receivers
are too slow to drain their receive queue, or by allowing
a bigger RWIN when MSS is close to PAGE_SIZE.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Link: https://lore.kernel.org/r/20230717152917.751987-1-edumazet@google.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2023-07-17 15:29:17 +00:00
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2023-07-20 20:47:50 +02:00
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tcp: increase the default TCP scaling ratio
After commit dfa2f0483360 ("tcp: get rid of sysctl_tcp_adv_win_scale"),
we noticed an application-level timeout due to reduced throughput.
Before the commit, for a client that sets SO_RCVBUF to 65k, it takes
around 22 seconds to transfer 10M data. After the commit, it takes 40
seconds. Because our application has a 30-second timeout, this
regression broke the application.
The reason that it takes longer to transfer data is that
tp->scaling_ratio is initialized to a value that results in ~0.25 of
rcvbuf. In our case, SO_RCVBUF is set to 65536 by the application, which
translates to 2 * 65536 = 131,072 bytes in rcvbuf and hence a ~28k
initial receive window.
Later, even though the scaling_ratio is updated to a more accurate
skb->len/skb->truesize, which is ~0.66 in our environment, the window
stays at ~0.25 * rcvbuf. This is because tp->window_clamp does not
change together with the tp->scaling_ratio update when autotuning is
disabled due to SO_RCVBUF. As a result, the window size is capped at the
initial window_clamp, which is also ~0.25 * rcvbuf, and never grows
bigger.
Most modern applications let the kernel do autotuning, and benefit from
the increased scaling_ratio. But there are applications such as kafka
that has a default setting of SO_RCVBUF=64k.
This patch increases the initial scaling_ratio from ~25% to 50% in order
to make it backward compatible with the original default
sysctl_tcp_adv_win_scale for applications setting SO_RCVBUF.
Fixes: dfa2f0483360 ("tcp: get rid of sysctl_tcp_adv_win_scale")
Signed-off-by: Hechao Li <hli@netflix.com>
Reviewed-by: Tycho Andersen <tycho@tycho.pizza>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://lore.kernel.org/netdev/20240402215405.432863-1-hli@netflix.com/
Signed-off-by: David S. Miller <davem@davemloft.net>
2024-04-09 09:43:55 -07:00
|
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2023-10-23 13:44:37 -07:00
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|
tcp: increase the default TCP scaling ratio
After commit dfa2f0483360 ("tcp: get rid of sysctl_tcp_adv_win_scale"),
we noticed an application-level timeout due to reduced throughput.
Before the commit, for a client that sets SO_RCVBUF to 65k, it takes
around 22 seconds to transfer 10M data. After the commit, it takes 40
seconds. Because our application has a 30-second timeout, this
regression broke the application.
The reason that it takes longer to transfer data is that
tp->scaling_ratio is initialized to a value that results in ~0.25 of
rcvbuf. In our case, SO_RCVBUF is set to 65536 by the application, which
translates to 2 * 65536 = 131,072 bytes in rcvbuf and hence a ~28k
initial receive window.
Later, even though the scaling_ratio is updated to a more accurate
skb->len/skb->truesize, which is ~0.66 in our environment, the window
stays at ~0.25 * rcvbuf. This is because tp->window_clamp does not
change together with the tp->scaling_ratio update when autotuning is
disabled due to SO_RCVBUF. As a result, the window size is capped at the
initial window_clamp, which is also ~0.25 * rcvbuf, and never grows
bigger.
Most modern applications let the kernel do autotuning, and benefit from
the increased scaling_ratio. But there are applications such as kafka
that has a default setting of SO_RCVBUF=64k.
This patch increases the initial scaling_ratio from ~25% to 50% in order
to make it backward compatible with the original default
sysctl_tcp_adv_win_scale for applications setting SO_RCVBUF.
Fixes: dfa2f0483360 ("tcp: get rid of sysctl_tcp_adv_win_scale")
Signed-off-by: Hechao Li <hli@netflix.com>
Reviewed-by: Tycho Andersen <tycho@tycho.pizza>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://lore.kernel.org/netdev/20240402215405.432863-1-hli@netflix.com/
Signed-off-by: David S. Miller <davem@davemloft.net>
2024-04-09 09:43:55 -07:00
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2023-10-23 13:44:37 -07:00
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tcp: get rid of sysctl_tcp_adv_win_scale
With modern NIC drivers shifting to full page allocations per
received frame, we face the following issue:
TCP has one per-netns sysctl used to tweak how to translate
a memory use into an expected payload (RWIN), in RX path.
tcp_win_from_space() implementation is limited to few cases.
For hosts dealing with various MSS, we either under estimate
or over estimate the RWIN we send to the remote peers.
For instance with the default sysctl_tcp_adv_win_scale value,
we expect to store 50% of payload per allocated chunk of memory.
For the typical use of MTU=1500 traffic, and order-0 pages allocations
by NIC drivers, we are sending too big RWIN, leading to potential
tcp collapse operations, which are extremely expensive and source
of latency spikes.
This patch makes sysctl_tcp_adv_win_scale obsolete, and instead
uses a per socket scaling factor, so that we can precisely
adjust the RWIN based on effective skb->len/skb->truesize ratio.
This patch alone can double TCP receive performance when receivers
are too slow to drain their receive queue, or by allowing
a bigger RWIN when MSS is close to PAGE_SIZE.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Link: https://lore.kernel.org/r/20230717152917.751987-1-edumazet@google.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2023-07-17 15:29:17 +00:00
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2023-10-23 13:44:37 -07:00
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2005-04-16 15:20:36 -07:00
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2014-10-20 18:15:50 +09:00
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2005-04-16 15:20:36 -07:00
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2019-10-10 20:17:44 -07:00
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2019-10-09 15:41:03 -07:00
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2005-04-16 15:20:36 -07:00
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2014-10-20 18:15:50 +09:00
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2005-04-16 15:20:36 -07:00
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2019-10-10 20:17:44 -07:00
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2005-04-16 15:20:36 -07:00
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2023-12-04 17:08:05 +01:00
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2021-09-29 10:25:13 -07:00
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2023-12-04 17:08:05 +01:00
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2021-09-29 10:25:13 -07:00
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2023-12-04 17:08:05 +01:00
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2020-09-14 10:01:18 +02:00
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2023-05-22 19:56:12 -07:00
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2020-09-14 10:01:18 +02:00
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2020-05-12 06:54:30 -07:00
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tcp: fix SO_RCVLOWAT related hangs under mem pressure
While commit 24adbc1676af ("tcp: fix SO_RCVLOWAT hangs with fat skbs")
fixed an issue vs too small sk_rcvbuf for given sk_rcvlowat constraint,
it missed to address issue caused by memory pressure.
1) If we are under memory pressure and socket receive queue is empty.
First incoming packet is allowed to be queued, after commit
76dfa6082032 ("tcp: allow one skb to be received per socket under memory pressure")
But we do not send EPOLLIN yet, in case tcp_data_ready() sees sk_rcvlowat
is bigger than skb length.
2) Then, when next packet comes, it is dropped, and we directly
call sk->sk_data_ready().
3) If application is using poll(), tcp_poll() will then use
tcp_stream_is_readable() and decide the socket receive queue is
not yet filled, so nothing will happen.
Even when sender retransmits packets, phases 2) & 3) repeat
and flow is effectively frozen, until memory pressure is off.
Fix is to consider tcp_under_memory_pressure() to take care
of global memory pressure or memcg pressure.
Fixes: 24adbc1676af ("tcp: fix SO_RCVLOWAT hangs with fat skbs")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Reported-by: Arjun Roy <arjunroy@google.com>
Suggested-by: Wei Wang <weiwan@google.com>
Reviewed-by: Wei Wang <weiwan@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2021-02-12 15:22:13 -08:00
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2020-05-12 06:54:30 -07:00
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2021-02-12 15:22:14 -08:00
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2014-05-11 20:22:11 -07:00
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2015-09-25 07:39:09 -07:00
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2014-05-11 20:22:11 -07:00
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2013-09-23 11:33:32 -07:00
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2017-06-07 13:29:12 -07:00
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2005-04-16 15:20:36 -07:00
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2016-01-07 16:38:45 +02:00
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2023-07-19 21:28:50 +00:00
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2016-01-07 16:38:45 +02:00
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2023-07-19 21:28:50 +00:00
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2005-04-16 15:20:36 -07:00
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2016-01-07 16:38:43 +02:00
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2023-07-19 21:28:49 +00:00
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2016-01-07 16:38:43 +02:00
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2023-07-19 21:28:49 +00:00
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2005-04-16 15:20:36 -07:00
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2009-08-28 23:48:54 -07:00
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2016-01-07 16:38:44 +02:00
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2023-07-19 21:28:51 +00:00
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2016-01-07 16:38:44 +02:00
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2023-07-19 21:28:51 +00:00
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2009-08-28 23:48:54 -07:00
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2010-04-26 18:33:27 +00:00
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2017-05-16 14:00:07 -07:00
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2010-04-26 18:33:27 +00:00
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2005-08-09 20:10:42 -07:00
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2005-04-16 15:20:36 -07:00
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2022-07-15 10:17:50 -07:00
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2005-08-09 20:10:42 -07:00
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2005-04-16 15:20:36 -07:00
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2005-08-09 20:10:42 -07:00
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2005-04-16 15:20:36 -07:00
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2012-05-16 23:15:34 +00:00
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2005-04-16 15:20:36 -07:00
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2009-03-14 14:23:03 +00:00
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2012-05-16 23:15:34 +00:00
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2018-07-11 12:16:12 +02:00
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2023-10-20 12:57:46 +00:00
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2012-05-16 23:15:34 +00:00
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2010-12-16 14:08:34 -08:00
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2012-05-16 23:15:34 +00:00
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2009-03-14 14:23:03 +00:00
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2012-05-16 23:15:34 +00:00
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2009-03-14 14:23:03 +00:00
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2012-05-16 23:15:34 +00:00
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2005-04-16 15:20:36 -07:00
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2018-07-11 12:16:12 +02:00
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2012-05-16 23:15:34 +00:00
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2005-04-16 15:20:36 -07:00
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2025-09-11 13:06:30 +02:00
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tcp: AccECN core
This change implements Accurate ECN without negotiation and
AccECN Option (that will be added by later changes). Based on
AccECN specifications:
https://tools.ietf.org/id/draft-ietf-tcpm-accurate-ecn-28.txt
Accurate ECN allows feeding back the number of CE (congestion
experienced) marks accurately to the sender in contrast to
RFC3168 ECN that can only signal one marks-seen-yes/no per RTT.
Congestion control algorithms can take advantage of the accurate
ECN information to fine-tune their congestion response to avoid
drastic rate reduction when only mild congestion is encountered.
With Accurate ECN, tp->received_ce (r.cep in AccECN spec) keeps
track of how many segments have arrived with a CE mark. Accurate
ECN uses ACE field (ECE, CWR, AE) to communicate the value back
to the sender which updates tp->delivered_ce (s.cep) based on the
feedback. This signalling channel is lossy when ACE field overflow
occurs.
Conservative strategy is selected here to deal with the ACE
overflow, however, some strategies using the AccECN option later
in the overall patchset mitigate against false overflows detected.
The ACE field values on the wire are offset by
TCP_ACCECN_CEP_INIT_OFFSET. Delivered_ce/received_ce count the
real CE marks rather than forcing all downstream users to adapt
to the wire offset.
This patch uses the first 1-byte hole and the last 4-byte hole of
the tcp_sock_write_txrx for 'received_ce_pending' and 'received_ce'.
Also, the group size of tcp_sock_write_txrx is increased from
91 + 4 to 95 + 4 due to the new u32 received_ce member. Below are
the trimmed pahole outcomes before and after this patch.
[BEFORE THIS PATCH]
struct tcp_sock {
[...]
__cacheline_group_begin__tcp_sock_write_txrx[0]; /* 2521 0 */
u8 nonagle:4; /* 2521: 0 1 */
u8 rate_app_limited:1; /* 2521: 4 1 */
/* XXX 3 bits hole, try to pack */
/* XXX 2 bytes hole, try to pack */
[...]
u32 delivered_ce; /* 2576 4 */
u32 app_limited; /* 2580 4 */
u32 rcv_wnd; /* 2684 4 */
struct tcp_options_received rx_opt; /* 2688 24 */
__cacheline_group_end__tcp_sock_write_txrx[0]; /* 2612 0 */
/* XXX 4 bytes hole, try to pack */
[...]
/* size: 3200, cachelines: 50, members: 161 */
}
[AFTER THIS PATCH]
struct tcp_sock {
[...]
__cacheline_group_begin__tcp_sock_write_txrx[0]; /* 2521 0 */
u8 nonagle:4; /* 2521: 0 1 */
u8 rate_app_limited:1; /* 2521: 4 1 */
/* XXX 3 bits hole, try to pack */
/* Force alignment to the next boundary: */
u8 :0;
u8 received_ce_pending:4;/* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
/* XXX 1 byte hole, try to pack */
[...]
u32 delivered_ce; /* 2576 4 */
u32 received_ce; /* 2580 4 */
u32 app_limited; /* 2584 4 */
u32 rcv_wnd; /* 2588 4 */
struct tcp_options_received rx_opt; /* 2592 24 */
__cacheline_group_end__tcp_sock_write_txrx[0]; /* 2616 0 */
[...]
/* size: 3200, cachelines: 50, members: 164 */
}
Signed-off-by: Ilpo Järvinen <ij@kernel.org>
Co-developed-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Signed-off-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Co-developed-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Signed-off-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250916082434.100722-2-chia-yu.chang@nokia-bell-labs.com
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-09-16 10:24:25 +02:00
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2025-09-11 13:06:30 +02:00
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|
tcp: AccECN core
This change implements Accurate ECN without negotiation and
AccECN Option (that will be added by later changes). Based on
AccECN specifications:
https://tools.ietf.org/id/draft-ietf-tcpm-accurate-ecn-28.txt
Accurate ECN allows feeding back the number of CE (congestion
experienced) marks accurately to the sender in contrast to
RFC3168 ECN that can only signal one marks-seen-yes/no per RTT.
Congestion control algorithms can take advantage of the accurate
ECN information to fine-tune their congestion response to avoid
drastic rate reduction when only mild congestion is encountered.
With Accurate ECN, tp->received_ce (r.cep in AccECN spec) keeps
track of how many segments have arrived with a CE mark. Accurate
ECN uses ACE field (ECE, CWR, AE) to communicate the value back
to the sender which updates tp->delivered_ce (s.cep) based on the
feedback. This signalling channel is lossy when ACE field overflow
occurs.
Conservative strategy is selected here to deal with the ACE
overflow, however, some strategies using the AccECN option later
in the overall patchset mitigate against false overflows detected.
The ACE field values on the wire are offset by
TCP_ACCECN_CEP_INIT_OFFSET. Delivered_ce/received_ce count the
real CE marks rather than forcing all downstream users to adapt
to the wire offset.
This patch uses the first 1-byte hole and the last 4-byte hole of
the tcp_sock_write_txrx for 'received_ce_pending' and 'received_ce'.
Also, the group size of tcp_sock_write_txrx is increased from
91 + 4 to 95 + 4 due to the new u32 received_ce member. Below are
the trimmed pahole outcomes before and after this patch.
[BEFORE THIS PATCH]
struct tcp_sock {
[...]
__cacheline_group_begin__tcp_sock_write_txrx[0]; /* 2521 0 */
u8 nonagle:4; /* 2521: 0 1 */
u8 rate_app_limited:1; /* 2521: 4 1 */
/* XXX 3 bits hole, try to pack */
/* XXX 2 bytes hole, try to pack */
[...]
u32 delivered_ce; /* 2576 4 */
u32 app_limited; /* 2580 4 */
u32 rcv_wnd; /* 2684 4 */
struct tcp_options_received rx_opt; /* 2688 24 */
__cacheline_group_end__tcp_sock_write_txrx[0]; /* 2612 0 */
/* XXX 4 bytes hole, try to pack */
[...]
/* size: 3200, cachelines: 50, members: 161 */
}
[AFTER THIS PATCH]
struct tcp_sock {
[...]
__cacheline_group_begin__tcp_sock_write_txrx[0]; /* 2521 0 */
u8 nonagle:4; /* 2521: 0 1 */
u8 rate_app_limited:1; /* 2521: 4 1 */
/* XXX 3 bits hole, try to pack */
/* Force alignment to the next boundary: */
u8 :0;
u8 received_ce_pending:4;/* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
/* XXX 1 byte hole, try to pack */
[...]
u32 delivered_ce; /* 2576 4 */
u32 received_ce; /* 2580 4 */
u32 app_limited; /* 2584 4 */
u32 rcv_wnd; /* 2588 4 */
struct tcp_options_received rx_opt; /* 2592 24 */
__cacheline_group_end__tcp_sock_write_txrx[0]; /* 2616 0 */
[...]
/* size: 3200, cachelines: 50, members: 164 */
}
Signed-off-by: Ilpo Järvinen <ij@kernel.org>
Co-developed-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Signed-off-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Co-developed-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Signed-off-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250916082434.100722-2-chia-yu.chang@nokia-bell-labs.com
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-09-16 10:24:25 +02:00
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2025-09-11 13:06:30 +02:00
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|
tcp: AccECN core
This change implements Accurate ECN without negotiation and
AccECN Option (that will be added by later changes). Based on
AccECN specifications:
https://tools.ietf.org/id/draft-ietf-tcpm-accurate-ecn-28.txt
Accurate ECN allows feeding back the number of CE (congestion
experienced) marks accurately to the sender in contrast to
RFC3168 ECN that can only signal one marks-seen-yes/no per RTT.
Congestion control algorithms can take advantage of the accurate
ECN information to fine-tune their congestion response to avoid
drastic rate reduction when only mild congestion is encountered.
With Accurate ECN, tp->received_ce (r.cep in AccECN spec) keeps
track of how many segments have arrived with a CE mark. Accurate
ECN uses ACE field (ECE, CWR, AE) to communicate the value back
to the sender which updates tp->delivered_ce (s.cep) based on the
feedback. This signalling channel is lossy when ACE field overflow
occurs.
Conservative strategy is selected here to deal with the ACE
overflow, however, some strategies using the AccECN option later
in the overall patchset mitigate against false overflows detected.
The ACE field values on the wire are offset by
TCP_ACCECN_CEP_INIT_OFFSET. Delivered_ce/received_ce count the
real CE marks rather than forcing all downstream users to adapt
to the wire offset.
This patch uses the first 1-byte hole and the last 4-byte hole of
the tcp_sock_write_txrx for 'received_ce_pending' and 'received_ce'.
Also, the group size of tcp_sock_write_txrx is increased from
91 + 4 to 95 + 4 due to the new u32 received_ce member. Below are
the trimmed pahole outcomes before and after this patch.
[BEFORE THIS PATCH]
struct tcp_sock {
[...]
__cacheline_group_begin__tcp_sock_write_txrx[0]; /* 2521 0 */
u8 nonagle:4; /* 2521: 0 1 */
u8 rate_app_limited:1; /* 2521: 4 1 */
/* XXX 3 bits hole, try to pack */
/* XXX 2 bytes hole, try to pack */
[...]
u32 delivered_ce; /* 2576 4 */
u32 app_limited; /* 2580 4 */
u32 rcv_wnd; /* 2684 4 */
struct tcp_options_received rx_opt; /* 2688 24 */
__cacheline_group_end__tcp_sock_write_txrx[0]; /* 2612 0 */
/* XXX 4 bytes hole, try to pack */
[...]
/* size: 3200, cachelines: 50, members: 161 */
}
[AFTER THIS PATCH]
struct tcp_sock {
[...]
__cacheline_group_begin__tcp_sock_write_txrx[0]; /* 2521 0 */
u8 nonagle:4; /* 2521: 0 1 */
u8 rate_app_limited:1; /* 2521: 4 1 */
/* XXX 3 bits hole, try to pack */
/* Force alignment to the next boundary: */
u8 :0;
u8 received_ce_pending:4;/* 2522: 0 1 */
u8 unused2:4; /* 2522: 4 1 */
/* XXX 1 byte hole, try to pack */
[...]
u32 delivered_ce; /* 2576 4 */
u32 received_ce; /* 2580 4 */
u32 app_limited; /* 2584 4 */
u32 rcv_wnd; /* 2588 4 */
struct tcp_options_received rx_opt; /* 2592 24 */
__cacheline_group_end__tcp_sock_write_txrx[0]; /* 2616 0 */
[...]
/* size: 3200, cachelines: 50, members: 164 */
}
Signed-off-by: Ilpo Järvinen <ij@kernel.org>
Co-developed-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Signed-off-by: Olivier Tilmans <olivier.tilmans@nokia.com>
Co-developed-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Signed-off-by: Chia-Yu Chang <chia-yu.chang@nokia-bell-labs.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20250916082434.100722-2-chia-yu.chang@nokia-bell-labs.com
Signed-off-by: Paolo Abeni <pabeni@redhat.com>
2025-09-16 10:24:25 +02:00
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2025-09-11 13:06:30 +02:00
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2015-03-16 21:06:20 -07:00
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2026-07-17 08:14:42 +00:00
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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>
2015-02-06 16:04:38 -05:00
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2008-07-16 20:21:42 -07:00
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2005-04-16 15:20:36 -07:00
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net: snmp: kill various STATS_USER() helpers
In the old days (before linux-3.0), SNMP counters were duplicated,
one for user context, and one for BH context.
After commit 8f0ea0fe3a03 ("snmp: reduce percpu needs by 50%")
we have a single copy, and what really matters is preemption being
enabled or disabled, since we use this_cpu_inc() or __this_cpu_inc()
respectively.
We therefore kill SNMP_INC_STATS_USER(), SNMP_ADD_STATS_USER(),
NET_INC_STATS_USER(), NET_ADD_STATS_USER(), SCTP_INC_STATS_USER(),
SNMP_INC_STATS64_USER(), SNMP_ADD_STATS64_USER(), TCP_ADD_STATS_USER(),
UDP_INC_STATS_USER(), UDP6_INC_STATS_USER(), and XFRM_INC_STATS_USER()
Following patches will rename __BH helpers to make clear their
usage is not tied to BH being disabled.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2016-04-27 16:44:27 -07:00
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2005-04-16 15:20:36 -07:00
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2007-09-20 11:30:48 -07:00
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2008-09-20 21:25:15 -07:00
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2005-11-10 17:14:59 -08:00
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2007-09-20 11:37:19 -07:00
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2023-10-23 20:21:54 +01:00
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2012-01-31 05:18:33 +00:00
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2006-11-14 19:07:45 -08:00
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2012-01-31 05:18:33 +00:00
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2006-11-14 19:07:45 -08:00
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2012-01-31 05:18:33 +00:00
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2017-06-15 18:07:06 -07:00
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2021-10-15 10:26:05 +03:00
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2019-12-30 14:14:28 -08:00
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2012-01-31 05:18:33 +00:00
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2006-11-14 19:07:45 -08:00
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2012-01-31 05:18:33 +00:00
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2012-01-31 18:45:40 +00:00
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2006-11-14 19:07:45 -08:00
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tcp: Convert tcp-md5 to use MD5 library instead of crypto_ahash
Make tcp-md5 use the MD5 library API (added in 6.18) instead of the
crypto_ahash API. This is much simpler and also more efficient:
- The library API just operates on struct md5_ctx. Just allocate this
struct on the stack instead of using a pool of pre-allocated
crypto_ahash and ahash_request objects.
- The library API accepts standard pointers and doesn't require
scatterlists. So, for hashing the headers just use an on-stack buffer
instead of a pool of pre-allocated kmalloc'ed scratch buffers.
- The library API never fails. Therefore, checking for MD5 hashing
errors is no longer necessary. Update tcp_v4_md5_hash_skb(),
tcp_v6_md5_hash_skb(), tcp_v4_md5_hash_hdr(), tcp_v6_md5_hash_hdr(),
tcp_md5_hash_key(), tcp_sock_af_ops::calc_md5_hash, and
tcp_request_sock_ops::calc_md5_hash to return void instead of int.
- The library API provides direct access to the MD5 code, eliminating
unnecessary overhead such as indirect function calls and scatterlist
management. Microbenchmarks of tcp_v4_md5_hash_skb() on x86_64 show a
speedup from 7518 to 7041 cycles (6% fewer) with skb->len == 1440, or
from 1020 to 678 cycles (33% fewer) with skb->len == 140.
Since tcp_sigpool_hash_skb_data() can no longer be used, add a function
tcp_md5_hash_skb_data() which is specialized to MD5. Of course, to the
extent that this duplicates any code, it's well worth it.
To preserve the existing behavior of TCP-MD5 support being disabled when
the kernel is booted with "fips=1", make tcp_md5_do_add() check
fips_enabled itself. Previously it relied on the error from
crypto_alloc_ahash("md5") being bubbled up. I don't know for sure that
this is actually needed, but this preserves the existing behavior.
Tested with bidirectional TCP-MD5, both IPv4 and IPv6, between a kernel
that includes this commit and a kernel that doesn't include this commit.
(Side note: please don't use TCP-MD5! It's cryptographically weak. But
as long as Linux supports it, it might as well be implemented properly.)
Signed-off-by: Eric Biggers <ebiggers@kernel.org>
Link: https://patch.msgid.link/20251014215836.115616-1-ebiggers@kernel.org
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2025-10-14 14:58:36 -07:00
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2013-09-23 11:33:32 -07:00
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2021-10-15 10:26:05 +03:00
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2022-11-23 17:38:57 +00:00
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2013-09-23 11:33:32 -07:00
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2021-10-15 10:26:05 +03:00
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2023-10-23 20:21:56 +01:00
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2015-09-25 07:39:15 -07:00
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2015-03-24 15:58:56 -07:00
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2006-11-14 19:07:45 -08:00
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2008-04-18 12:45:16 +09:00
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2019-12-30 14:14:28 -08:00
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2018-11-27 15:03:21 -08:00
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2023-10-23 20:21:56 +01:00
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2018-11-27 15:03:21 -08:00
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2019-12-30 14:14:28 -08:00
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2018-11-27 15:03:21 -08:00
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2022-11-23 17:38:57 +00:00
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2018-11-27 15:03:21 -08:00
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2023-10-23 20:21:56 +01:00
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2018-11-27 15:03:21 -08:00
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2022-03-07 16:44:21 -08:00
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2012-01-31 05:18:33 +00:00
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tcp: Destroy TCP-AO, TCP-MD5 keys in .sk_destruct()
Currently there are a couple of minor issues with destroying the keys
tcp_v4_destroy_sock():
1. The socket is yet in TCP bind buckets, making it reachable for
incoming segments [on another CPU core], potentially available to send
late FIN/ACK/RST replies.
2. There is at least one code path, where tcp_done() is called before
sending RST [kudos to Bob for investigation]. This is a case of
a server, that finished sending its data and just called close().
The socket is in TCP_FIN_WAIT2 and has RCV_SHUTDOWN (set by
__tcp_close())
tcp_v4_do_rcv()/tcp_v6_do_rcv()
tcp_rcv_state_process() /* LINUX_MIB_TCPABORTONDATA */
tcp_reset()
tcp_done_with_error()
tcp_done()
inet_csk_destroy_sock() /* Destroys AO/MD5 keys */
/* tcp_rcv_state_process() returns SKB_DROP_REASON_TCP_ABORT_ON_DATA */
tcp_v4_send_reset() /* Sends an unsigned RST segment */
tcpdump:
> 22:53:15.399377 00:00:b2:1f:00:00 > 00:00:01:01:00:00, ethertype IPv4 (0x0800), length 74: (tos 0x0, ttl 64, id 33929, offset 0, flags [DF], proto TCP (6), length 60)
> 1.0.0.1.34567 > 1.0.0.2.49848: Flags [F.], seq 2185658590, ack 3969644355, win 502, options [nop,nop,md5 valid], length 0
> 22:53:15.399396 00:00:01:01:00:00 > 00:00:b2:1f:00:00, ethertype IPv4 (0x0800), length 86: (tos 0x0, ttl 64, id 51951, offset 0, flags [DF], proto TCP (6), length 72)
> 1.0.0.2.49848 > 1.0.0.1.34567: Flags [.], seq 3969644375, ack 2185658591, win 128, options [nop,nop,md5 valid,nop,nop,sack 1 {2185658590:2185658591}], length 0
> 22:53:16.429588 00:00:b2:1f:00:00 > 00:00:01:01:00:00, ethertype IPv4 (0x0800), length 60: (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto TCP (6), length 40)
> 1.0.0.1.34567 > 1.0.0.2.49848: Flags [R], seq 2185658590, win 0, length 0
> 22:53:16.664725 00:00:b2:1f:00:00 > 00:00:01:01:00:00, ethertype IPv4 (0x0800), length 74: (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto TCP (6), length 60)
> 1.0.0.1.34567 > 1.0.0.2.49848: Flags [R], seq 2185658591, win 0, options [nop,nop,md5 valid], length 0
> 22:53:17.289832 00:00:b2:1f:00:00 > 00:00:01:01:00:00, ethertype IPv4 (0x0800), length 74: (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto TCP (6), length 60)
> 1.0.0.1.34567 > 1.0.0.2.49848: Flags [R], seq 2185658591, win 0, options [nop,nop,md5 valid], length 0
Note the signed RSTs later in the dump - those are sent by the server
when the fin-wait socket gets removed from hash buckets, by
the listener socket.
Instead of destroying AO/MD5 info and their keys in inet_csk_destroy_sock(),
slightly delay it until the actual socket .sk_destruct(). As shutdown'ed
socket can yet send non-data replies, they should be signed in order for
the peer to process them. Now it also matches how AO/MD5 gets destructed
for TIME-WAIT sockets (in tcp_twsk_destructor()).
This seems optimal for TCP-MD5, while for TCP-AO it seems to have an
open problem: once RST get sent and socket gets actually destructed,
there is no information on the initial sequence numbers. So, in case
this last RST gets lost in the network, the server's listener socket
won't be able to properly sign another RST. Nothing in RFC 1122
prescribes keeping any local state after non-graceful reset.
Luckily, BGP are known to use keep alive(s).
While the issue is quite minor/cosmetic, these days monitoring network
counters is a common practice and getting invalid signed segments from
a trusted BGP peer can get customers worried.
Investigated-by: Bob Gilligan <gilligan@arista.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Kuniyuki Iwashima <kuniyu@google.com>
Signed-off-by: Dmitry Safonov <dima@arista.com>
Link: https://patch.msgid.link/20250909-b4-tcp-ao-md5-rst-finwait2-v5-1-9ffaaaf8b236@arista.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2025-09-09 02:18:50 +01:00
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2008-04-18 12:45:16 +09:00
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2019-12-30 14:14:28 -08:00
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2012-01-31 05:18:33 +00:00
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2022-03-07 16:44:21 -08:00
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2023-10-23 20:21:56 +01:00
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2008-04-18 12:45:16 +09:00
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tcp: Destroy TCP-AO, TCP-MD5 keys in .sk_destruct()
Currently there are a couple of minor issues with destroying the keys
tcp_v4_destroy_sock():
1. The socket is yet in TCP bind buckets, making it reachable for
incoming segments [on another CPU core], potentially available to send
late FIN/ACK/RST replies.
2. There is at least one code path, where tcp_done() is called before
sending RST [kudos to Bob for investigation]. This is a case of
a server, that finished sending its data and just called close().
The socket is in TCP_FIN_WAIT2 and has RCV_SHUTDOWN (set by
__tcp_close())
tcp_v4_do_rcv()/tcp_v6_do_rcv()
tcp_rcv_state_process() /* LINUX_MIB_TCPABORTONDATA */
tcp_reset()
tcp_done_with_error()
tcp_done()
inet_csk_destroy_sock() /* Destroys AO/MD5 keys */
/* tcp_rcv_state_process() returns SKB_DROP_REASON_TCP_ABORT_ON_DATA */
tcp_v4_send_reset() /* Sends an unsigned RST segment */
tcpdump:
> 22:53:15.399377 00:00:b2:1f:00:00 > 00:00:01:01:00:00, ethertype IPv4 (0x0800), length 74: (tos 0x0, ttl 64, id 33929, offset 0, flags [DF], proto TCP (6), length 60)
> 1.0.0.1.34567 > 1.0.0.2.49848: Flags [F.], seq 2185658590, ack 3969644355, win 502, options [nop,nop,md5 valid], length 0
> 22:53:15.399396 00:00:01:01:00:00 > 00:00:b2:1f:00:00, ethertype IPv4 (0x0800), length 86: (tos 0x0, ttl 64, id 51951, offset 0, flags [DF], proto TCP (6), length 72)
> 1.0.0.2.49848 > 1.0.0.1.34567: Flags [.], seq 3969644375, ack 2185658591, win 128, options [nop,nop,md5 valid,nop,nop,sack 1 {2185658590:2185658591}], length 0
> 22:53:16.429588 00:00:b2:1f:00:00 > 00:00:01:01:00:00, ethertype IPv4 (0x0800), length 60: (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto TCP (6), length 40)
> 1.0.0.1.34567 > 1.0.0.2.49848: Flags [R], seq 2185658590, win 0, length 0
> 22:53:16.664725 00:00:b2:1f:00:00 > 00:00:01:01:00:00, ethertype IPv4 (0x0800), length 74: (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto TCP (6), length 60)
> 1.0.0.1.34567 > 1.0.0.2.49848: Flags [R], seq 2185658591, win 0, options [nop,nop,md5 valid], length 0
> 22:53:17.289832 00:00:b2:1f:00:00 > 00:00:01:01:00:00, ethertype IPv4 (0x0800), length 74: (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto TCP (6), length 60)
> 1.0.0.1.34567 > 1.0.0.2.49848: Flags [R], seq 2185658591, win 0, options [nop,nop,md5 valid], length 0
Note the signed RSTs later in the dump - those are sent by the server
when the fin-wait socket gets removed from hash buckets, by
the listener socket.
Instead of destroying AO/MD5 info and their keys in inet_csk_destroy_sock(),
slightly delay it until the actual socket .sk_destruct(). As shutdown'ed
socket can yet send non-data replies, they should be signed in order for
the peer to process them. Now it also matches how AO/MD5 gets destructed
for TIME-WAIT sockets (in tcp_twsk_destructor()).
This seems optimal for TCP-MD5, while for TCP-AO it seems to have an
open problem: once RST get sent and socket gets actually destructed,
there is no information on the initial sequence numbers. So, in case
this last RST gets lost in the network, the server's listener socket
won't be able to properly sign another RST. Nothing in RFC 1122
prescribes keeping any local state after non-graceful reset.
Luckily, BGP are known to use keep alive(s).
While the issue is quite minor/cosmetic, these days monitoring network
counters is a common practice and getting invalid signed segments from
a trusted BGP peer can get customers worried.
Investigated-by: Bob Gilligan <gilligan@arista.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Kuniyuki Iwashima <kuniyu@google.com>
Signed-off-by: Dmitry Safonov <dima@arista.com>
Link: https://patch.msgid.link/20250909-b4-tcp-ao-md5-rst-finwait2-v5-1-9ffaaaf8b236@arista.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2025-09-09 02:18:50 +01:00
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2008-04-18 12:45:16 +09:00
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tcp: Convert tcp-md5 to use MD5 library instead of crypto_ahash
Make tcp-md5 use the MD5 library API (added in 6.18) instead of the
crypto_ahash API. This is much simpler and also more efficient:
- The library API just operates on struct md5_ctx. Just allocate this
struct on the stack instead of using a pool of pre-allocated
crypto_ahash and ahash_request objects.
- The library API accepts standard pointers and doesn't require
scatterlists. So, for hashing the headers just use an on-stack buffer
instead of a pool of pre-allocated kmalloc'ed scratch buffers.
- The library API never fails. Therefore, checking for MD5 hashing
errors is no longer necessary. Update tcp_v4_md5_hash_skb(),
tcp_v6_md5_hash_skb(), tcp_v4_md5_hash_hdr(), tcp_v6_md5_hash_hdr(),
tcp_md5_hash_key(), tcp_sock_af_ops::calc_md5_hash, and
tcp_request_sock_ops::calc_md5_hash to return void instead of int.
- The library API provides direct access to the MD5 code, eliminating
unnecessary overhead such as indirect function calls and scatterlist
management. Microbenchmarks of tcp_v4_md5_hash_skb() on x86_64 show a
speedup from 7518 to 7041 cycles (6% fewer) with skb->len == 1440, or
from 1020 to 678 cycles (33% fewer) with skb->len == 140.
Since tcp_sigpool_hash_skb_data() can no longer be used, add a function
tcp_md5_hash_skb_data() which is specialized to MD5. Of course, to the
extent that this duplicates any code, it's well worth it.
To preserve the existing behavior of TCP-MD5 support being disabled when
the kernel is booted with "fips=1", make tcp_md5_do_add() check
fips_enabled itself. Previously it relied on the error from
crypto_alloc_ahash("md5") being bubbled up. I don't know for sure that
this is actually needed, but this preserves the existing behavior.
Tested with bidirectional TCP-MD5, both IPv4 and IPv6, between a kernel
that includes this commit and a kernel that doesn't include this commit.
(Side note: please don't use TCP-MD5! It's cryptographically weak. But
as long as Linux supports it, it might as well be implemented properly.)
Signed-off-by: Eric Biggers <ebiggers@kernel.org>
Link: https://patch.msgid.link/20251014215836.115616-1-ebiggers@kernel.org
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2025-10-14 14:58:36 -07:00
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2006-11-14 19:07:45 -08:00
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2012-08-31 12:29:11 +00:00
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2013-09-23 11:33:32 -07:00
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2017-12-12 13:10:40 -08:00
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2013-09-23 11:33:32 -07:00
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2015-04-06 14:37:27 -07:00
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2012-07-19 06:43:07 +00:00
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2014-02-20 10:09:18 -08:00
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2019-01-25 11:17:23 -05:00
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2012-07-19 06:43:07 +00:00
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2017-10-18 11:22:51 -07:00
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2017-09-27 11:35:42 +08:00
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2017-10-18 11:22:51 -07:00
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2019-06-19 23:46:28 +02:00
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2020-08-10 13:38:39 -04:00
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2016-02-01 21:03:07 -08:00
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2015-09-24 17:16:05 -07:00
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2017-10-23 13:22:23 -07:00
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2017-09-27 11:35:42 +08:00
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2017-01-23 10:59:20 -08:00
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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>
2017-01-23 10:59:22 -08:00
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2019-06-19 23:46:28 +02:00
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2019-05-29 12:33:57 -04:00
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2012-08-31 12:29:11 +00:00
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2019-06-19 23:46:28 +02:00
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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 10:09:33 +02:00
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2012-08-31 12:29:11 +00:00
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2017-04-20 14:45:47 -07:00
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|
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>
2017-04-20 14:45:46 -07:00
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|
|
2017-12-12 13:10:40 -08:00
|
|
|
|
net/tcp_fastopen: Disable active side TFO in certain scenarios
Middlebox firewall issues can potentially cause server's data being
blackholed after a successful 3WHS using TFO. Following are the related
reports from Apple:
https://www.nanog.org/sites/default/files/Paasch_Network_Support.pdf
Slide 31 identifies an issue where the client ACK to the server's data
sent during a TFO'd handshake is dropped.
C ---> syn-data ---> S
C <--- syn/ack ----- S
C (accept & write)
C <---- data ------- S
C ----- ACK -> X S
[retry and timeout]
https://www.ietf.org/proceedings/94/slides/slides-94-tcpm-13.pdf
Slide 5 shows a similar situation that the server's data gets dropped
after 3WHS.
C ---- syn-data ---> S
C <--- syn/ack ----- S
C ---- ack --------> S
S (accept & write)
C? X <- data ------ S
[retry and timeout]
This is the worst failure b/c the client can not detect such behavior to
mitigate the situation (such as disabling TFO). Failing to proceed, the
application (e.g., SSL library) may simply timeout and retry with TFO
again, and the process repeats indefinitely.
The proposed solution is to disable active TFO globally under the
following circumstances:
1. client side TFO socket detects out of order FIN
2. client side TFO socket receives out of order RST
We disable active side TFO globally for 1hr at first. Then if it
happens again, we disable it for 2h, then 4h, 8h, ...
And we reset the timeout to 1hr if a client side TFO sockets not opened
on loopback has successfully received data segs from server.
And we examine this condition during close().
The rational behind it is that when such firewall issue happens,
application running on the client should eventually close the socket as
it is not able to get the data it is expecting. Or application running
on the server should close the socket as it is not able to receive any
response from client.
In both cases, out of order FIN or RST will get received on the client
given that the firewall will not block them as no data are in those
frames.
And we want to disable active TFO globally as it helps if the middle box
is very close to the client and most of the connections are likely to
fail.
Also, add a debug sysctl:
tcp_fastopen_blackhole_detect_timeout_sec:
the initial timeout to use when firewall blackhole issue happens.
This can be set and read.
When setting it to 0, it means to disable the active disable logic.
Signed-off-by: Wei Wang <weiwan@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2017-04-20 14:45:46 -07:00
|
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|
2019-05-29 12:33:57 -04:00
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|
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 10:09:33 +02:00
|
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|
|
2019-05-29 12:33:57 -04:00
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2016-11-27 23:07:13 -08:00
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2026-03-08 12:35:49 +00:00
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2026-04-16 20:03:06 +00:00
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2026-03-08 12:35:49 +00:00
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2026-04-16 20:03:06 +00:00
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2026-03-08 12:35:49 +00:00
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2016-11-27 23:07:13 -08:00
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2017-10-04 12:59:58 -07:00
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2017-10-05 22:21:22 -07:00
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2007-03-07 12:12:44 -08:00
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2017-10-05 22:21:27 -07:00
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2019-07-19 11:52:33 -07:00
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2011-10-21 05:22:42 -04:00
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2007-03-07 12:12:44 -08:00
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2008-09-23 00:50:13 -07:00
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2007-03-07 12:12:44 -08:00
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2007-12-02 00:48:02 +02:00
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2008-09-23 00:50:13 -07:00
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2007-12-02 00:48:02 +02:00
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2011-10-21 05:22:42 -04:00
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2007-03-07 12:12:44 -08:00
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2017-10-05 22:21:27 -07:00
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2007-03-07 12:12:44 -08:00
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2008-09-23 00:50:13 -07:00
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2019-12-12 12:55:30 -08:00
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2017-10-05 22:21:27 -07:00
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2007-03-07 12:12:44 -08:00
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2019-12-12 12:55:30 -08:00
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2017-10-05 22:21:27 -07:00
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2007-03-07 12:12:44 -08:00
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2019-02-26 09:49:10 -08:00
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2007-03-07 12:12:44 -08:00
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2017-11-14 21:02:19 -08:00
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2016-11-27 23:07:14 -08:00
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2007-03-07 12:12:44 -08:00
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2008-09-21 21:28:51 -07:00
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2007-03-07 12:12:44 -08:00
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2008-09-21 21:28:51 -07:00
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2007-03-07 12:12:44 -08:00
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2017-10-11 13:27:29 -07:00
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2007-03-07 12:12:44 -08:00
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2017-10-05 22:21:27 -07:00
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2007-03-07 12:12:44 -08:00
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2017-10-05 22:21:27 -07:00
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2021-10-29 22:05:41 -04:00
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2007-03-07 12:12:44 -08:00
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2024-05-30 16:36:15 -07:00
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2009-12-08 22:26:13 +00:00
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2012-02-27 17:52:52 -05:00
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2007-11-15 19:41:46 -08:00
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2007-12-02 00:48:06 +02:00
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2007-11-15 19:41:46 -08:00
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2007-12-02 00:48:06 +02:00
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2017-11-14 21:02:19 -08:00
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2007-12-02 00:48:06 +02:00
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2017-11-14 21:02:19 -08:00
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2007-12-02 00:48:06 +02:00
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2017-10-30 23:08:20 -07:00
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2007-12-02 00:48:06 +02:00
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2017-10-30 23:08:20 -07:00
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2007-12-02 00:48:06 +02:00
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2015-12-21 21:29:24 +01:00
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2023-08-16 08:15:41 +00:00
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2015-12-21 21:29:24 +01:00
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2010-02-18 02:45:45 +00:00
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2012-05-16 23:15:34 +00:00
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2010-02-18 02:45:45 +00:00
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2005-04-16 15:20:36 -07:00
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2018-04-11 09:31:28 +02:00
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2011-10-30 06:46:30 +00:00
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2005-04-16 15:20:36 -07:00
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2011-10-30 06:46:30 +00:00
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2005-04-16 15:20:36 -07:00
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2008-04-13 22:11:14 -07:00
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2005-04-16 15:20:36 -07:00
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2012-05-24 01:10:10 -06:00
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2010-06-07 00:43:42 -07:00
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2005-04-16 15:20:36 -07:00
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2005-08-16 02:18:02 -03:00
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2008-02-07 21:49:26 -08:00
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2005-08-16 02:18:02 -03:00
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2013-09-23 11:33:32 -07:00
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2005-08-16 02:18:02 -03:00
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2013-10-18 10:36:17 -07:00
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2013-09-23 11:33:32 -07:00
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2024-05-02 10:44:45 +02:00
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2024-05-02 10:44:46 +02:00
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2020-06-23 15:31:14 -07:00
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net: stub tcp_gro_complete if CONFIG_INET=n
A few networking drivers including bnx2x, bnxt, qede, and idpf call
tcp_gro_complete as part of offloading TCP GRO. The function is only
defined if CONFIG_INET is true, since its TCP specific and is meaningless
if the kernel lacks IP networking support.
The combination of trying to use the complex network drivers with
CONFIG_NET but not CONFIG_INET is rather unlikely in practice: most use
cases are going to need IP networking.
The tcp_gro_complete function just sets some data in the socket buffer for
use in processing the TCP packet in the event that the GRO was offloaded to
the device. If the kernel lacks TCP support, such setup will simply go
unused.
The bnx2x, bnxt, and qede drivers wrap their TCP offload support in
CONFIG_INET checks and skip handling on such kernels.
The idpf driver did not check CONFIG_INET and thus fails to link if the
kernel is configured with CONFIG_NET=y, CONFIG_IDPF=(m|y), and
CONFIG_INET=n.
While checking CONFIG_INET does allow the driver to bypass significantly
more instructions in the event that we know TCP networking isn't supported,
the configuration is unlikely to be used widely.
Rather than require driver authors to care about this, stub the
tcp_gro_complete function when CONFIG_INET=n. This allows drivers to be
left as-is. It does mean the idpf driver will perform slightly more work
than strictly necessary when CONFIG_INET=n, since it will still execute
some of the skb setup in idpf_rx_rsc. However, that work would be performed
in the case where CONFIG_INET=y anyways.
I did not change the existing drivers, since they appear to wrap a
significant portion of code when CONFIG_INET=n. There is little benefit in
trashing these drivers just to unwrap and remove the CONFIG_INET check.
Using a stub for tcp_gro_complete is still beneficial, as it means future
drivers no longer need to worry about this case of CONFIG_NET=y and
CONFIG_INET=n, which should reduce noise from buildbots that check such a
configuration.
Signed-off-by: Jacob Keller <jacob.e.keller@intel.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Tested-by: Randy Dunlap <rdunlap@infradead.org> # build-tested
Link: https://lore.kernel.org/r/20231013185502.1473541-1-jacob.e.keller@intel.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2023-10-13 11:54:50 -07:00
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2023-05-29 16:44:30 +03:00
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|
net: stub tcp_gro_complete if CONFIG_INET=n
A few networking drivers including bnx2x, bnxt, qede, and idpf call
tcp_gro_complete as part of offloading TCP GRO. The function is only
defined if CONFIG_INET is true, since its TCP specific and is meaningless
if the kernel lacks IP networking support.
The combination of trying to use the complex network drivers with
CONFIG_NET but not CONFIG_INET is rather unlikely in practice: most use
cases are going to need IP networking.
The tcp_gro_complete function just sets some data in the socket buffer for
use in processing the TCP packet in the event that the GRO was offloaded to
the device. If the kernel lacks TCP support, such setup will simply go
unused.
The bnx2x, bnxt, and qede drivers wrap their TCP offload support in
CONFIG_INET checks and skip handling on such kernels.
The idpf driver did not check CONFIG_INET and thus fails to link if the
kernel is configured with CONFIG_NET=y, CONFIG_IDPF=(m|y), and
CONFIG_INET=n.
While checking CONFIG_INET does allow the driver to bypass significantly
more instructions in the event that we know TCP networking isn't supported,
the configuration is unlikely to be used widely.
Rather than require driver authors to care about this, stub the
tcp_gro_complete function when CONFIG_INET=n. This allows drivers to be
left as-is. It does mean the idpf driver will perform slightly more work
than strictly necessary when CONFIG_INET=n, since it will still execute
some of the skb setup in idpf_rx_rsc. However, that work would be performed
in the case where CONFIG_INET=y anyways.
I did not change the existing drivers, since they appear to wrap a
significant portion of code when CONFIG_INET=n. There is little benefit in
trashing these drivers just to unwrap and remove the CONFIG_INET check.
Using a stub for tcp_gro_complete is still beneficial, as it means future
drivers no longer need to worry about this case of CONFIG_NET=y and
CONFIG_INET=n, which should reduce noise from buildbots that check such a
configuration.
Signed-off-by: Jacob Keller <jacob.e.keller@intel.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Tested-by: Randy Dunlap <rdunlap@infradead.org> # build-tested
Link: https://lore.kernel.org/r/20231013185502.1473541-1-jacob.e.keller@intel.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2023-10-13 11:54:50 -07:00
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2013-06-07 05:11:46 +00:00
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2026-02-23 10:07:27 +00:00
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2006-06-22 03:02:40 -07:00
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|
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>
2013-07-22 20:27:07 -07:00
|
|
|
|
|
|
|
|
|
2016-02-03 09:46:57 +02:00
|
|
|
|
2023-07-19 21:28:55 +00:00
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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|
|
|
|
|
|
|
|
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>
2013-07-22 20:27:07 -07:00
|
|
|
|
|
|
|
|
|
2020-11-13 07:08:08 -08:00
|
|
|
|
tcp: TCP_NOTSENT_LOWAT socket option
Idea of this patch is to add optional limitation of number of
unsent bytes in TCP sockets, to reduce usage of kernel memory.
TCP receiver might announce a big window, and TCP sender autotuning
might allow a large amount of bytes in write queue, but this has little
performance impact if a large part of this buffering is wasted :
Write queue needs to be large only to deal with large BDP, not
necessarily to cope with scheduling delays (incoming ACKS make room
for the application to queue more bytes)
For most workloads, using a value of 128 KB or less is OK to give
applications enough time to react to POLLOUT events in time
(or being awaken in a blocking sendmsg())
This patch adds two ways to set the limit :
1) Per socket option TCP_NOTSENT_LOWAT
2) A sysctl (/proc/sys/net/ipv4/tcp_notsent_lowat) for sockets
not using TCP_NOTSENT_LOWAT socket option (or setting a zero value)
Default value being UINT_MAX (0xFFFFFFFF), meaning this has no effect.
This changes poll()/select()/epoll() to report POLLOUT
only if number of unsent bytes is below tp->nosent_lowat
Note this might increase number of sendmsg()/sendfile() calls
when using non blocking sockets,
and increase number of context switches for blocking sockets.
Note this is not related to SO_SNDLOWAT (as SO_SNDLOWAT is
defined as :
Specify the minimum number of bytes in the buffer until
the socket layer will pass the data to the protocol)
Tested:
netperf sessions, and watching /proc/net/protocols "memory" column for TCP
With 200 concurrent netperf -t TCP_STREAM sessions, amount of kernel memory
used by TCP buffers shrinks by ~55 % (20567 pages instead of 45458)
lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols
TCPv6 1880 2 45458 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y
TCP 1696 508 45458 no 208 yes kernel y y y y y y y y y y y y y n y y y y y
lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# (super_netperf 200 -t TCP_STREAM -H remote -l 90 &); sleep 60 ; grep TCP /proc/net/protocols
TCPv6 1880 2 20567 no 208 yes ipv6 y y y y y y y y y y y y y n y y y y y
TCP 1696 508 20567 no 208 yes kernel y y y y y y y y y y y y y n y y y y y
Using 128KB has no bad effect on the throughput or cpu usage
of a single flow, although there is an increase of context switches.
A bonus is that we hold socket lock for a shorter amount
of time and should improve latencies of ACK processing.
lpq83:~# echo -1 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3
OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf.
Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service
Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand
Size Size Size (sec) Util Util Util Util Demand Demand Units
Final Final % Method % Method
1651584 6291456 16384 20.00 17447.90 10^6bits/s 3.13 S -1.00 U 0.353 -1.000 usec/KB
Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3':
412,514 context-switches
200.034645535 seconds time elapsed
lpq83:~# echo 131072 >/proc/sys/net/ipv4/tcp_notsent_lowat
lpq83:~# perf stat -e context-switches ./netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3
OMNI Send TEST from 0.0.0.0 (0.0.0.0) port 0 AF_INET to 7.7.7.84 () port 0 AF_INET : +/-2.500% @ 99% conf.
Local Remote Local Elapsed Throughput Throughput Local Local Remote Remote Local Remote Service
Send Socket Recv Socket Send Time Units CPU CPU CPU CPU Service Service Demand
Size Size Size (sec) Util Util Util Util Demand Demand Units
Final Final % Method % Method
1593240 6291456 16384 20.00 17321.16 10^6bits/s 3.35 S -1.00 U 0.381 -1.000 usec/KB
Performance counter stats for './netperf -H 7.7.7.84 -t omni -l 20 -c -i10,3':
2,675,818 context-switches
200.029651391 seconds time elapsed
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-By: Yuchung Cheng <ycheng@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2013-07-22 20:27:07 -07:00
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2005-08-16 02:18:02 -03:00
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2013-09-23 11:33:32 -07:00
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2005-08-16 02:18:02 -03:00
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2015-09-25 07:39:23 -07:00
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2014-06-25 17:10:02 +03:00
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2014-06-25 17:09:59 +03:00
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2006-11-14 19:07:45 -08:00
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2015-09-25 07:39:15 -07:00
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2015-03-24 15:58:56 -07:00
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tcp: Convert tcp-md5 to use MD5 library instead of crypto_ahash
Make tcp-md5 use the MD5 library API (added in 6.18) instead of the
crypto_ahash API. This is much simpler and also more efficient:
- The library API just operates on struct md5_ctx. Just allocate this
struct on the stack instead of using a pool of pre-allocated
crypto_ahash and ahash_request objects.
- The library API accepts standard pointers and doesn't require
scatterlists. So, for hashing the headers just use an on-stack buffer
instead of a pool of pre-allocated kmalloc'ed scratch buffers.
- The library API never fails. Therefore, checking for MD5 hashing
errors is no longer necessary. Update tcp_v4_md5_hash_skb(),
tcp_v6_md5_hash_skb(), tcp_v4_md5_hash_hdr(), tcp_v6_md5_hash_hdr(),
tcp_md5_hash_key(), tcp_sock_af_ops::calc_md5_hash, and
tcp_request_sock_ops::calc_md5_hash to return void instead of int.
- The library API provides direct access to the MD5 code, eliminating
unnecessary overhead such as indirect function calls and scatterlist
management. Microbenchmarks of tcp_v4_md5_hash_skb() on x86_64 show a
speedup from 7518 to 7041 cycles (6% fewer) with skb->len == 1440, or
from 1020 to 678 cycles (33% fewer) with skb->len == 140.
Since tcp_sigpool_hash_skb_data() can no longer be used, add a function
tcp_md5_hash_skb_data() which is specialized to MD5. Of course, to the
extent that this duplicates any code, it's well worth it.
To preserve the existing behavior of TCP-MD5 support being disabled when
the kernel is booted with "fips=1", make tcp_md5_do_add() check
fips_enabled itself. Previously it relied on the error from
crypto_alloc_ahash("md5") being bubbled up. I don't know for sure that
this is actually needed, but this preserves the existing behavior.
Tested with bidirectional TCP-MD5, both IPv4 and IPv6, between a kernel
that includes this commit and a kernel that doesn't include this commit.
(Side note: please don't use TCP-MD5! It's cryptographically weak. But
as long as Linux supports it, it might as well be implemented properly.)
Signed-off-by: Eric Biggers <ebiggers@kernel.org>
Link: https://patch.msgid.link/20251014215836.115616-1-ebiggers@kernel.org
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2025-10-14 14:58:36 -07:00
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2015-03-24 15:58:55 -07:00
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2017-06-15 18:07:07 -07:00
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2020-07-23 08:09:05 +02:00
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2015-03-24 15:58:55 -07:00
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2006-11-14 19:07:45 -08:00
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2023-10-23 20:21:55 +01:00
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2023-10-23 20:21:56 +01:00
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2026-04-27 10:27:26 -07:00
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2023-10-23 20:21:58 +01:00
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2023-10-23 20:21:55 +01:00
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2006-11-14 19:07:45 -08:00
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2014-06-25 17:10:00 +03:00
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2006-11-14 19:07:45 -08:00
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2015-09-25 07:39:15 -07:00
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2015-03-24 15:58:56 -07:00
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tcp: Convert tcp-md5 to use MD5 library instead of crypto_ahash
Make tcp-md5 use the MD5 library API (added in 6.18) instead of the
crypto_ahash API. This is much simpler and also more efficient:
- The library API just operates on struct md5_ctx. Just allocate this
struct on the stack instead of using a pool of pre-allocated
crypto_ahash and ahash_request objects.
- The library API accepts standard pointers and doesn't require
scatterlists. So, for hashing the headers just use an on-stack buffer
instead of a pool of pre-allocated kmalloc'ed scratch buffers.
- The library API never fails. Therefore, checking for MD5 hashing
errors is no longer necessary. Update tcp_v4_md5_hash_skb(),
tcp_v6_md5_hash_skb(), tcp_v4_md5_hash_hdr(), tcp_v6_md5_hash_hdr(),
tcp_md5_hash_key(), tcp_sock_af_ops::calc_md5_hash, and
tcp_request_sock_ops::calc_md5_hash to return void instead of int.
- The library API provides direct access to the MD5 code, eliminating
unnecessary overhead such as indirect function calls and scatterlist
management. Microbenchmarks of tcp_v4_md5_hash_skb() on x86_64 show a
speedup from 7518 to 7041 cycles (6% fewer) with skb->len == 1440, or
from 1020 to 678 cycles (33% fewer) with skb->len == 140.
Since tcp_sigpool_hash_skb_data() can no longer be used, add a function
tcp_md5_hash_skb_data() which is specialized to MD5. Of course, to the
extent that this duplicates any code, it's well worth it.
To preserve the existing behavior of TCP-MD5 support being disabled when
the kernel is booted with "fips=1", make tcp_md5_do_add() check
fips_enabled itself. Previously it relied on the error from
crypto_alloc_ahash("md5") being bubbled up. I don't know for sure that
this is actually needed, but this preserves the existing behavior.
Tested with bidirectional TCP-MD5, both IPv4 and IPv6, between a kernel
that includes this commit and a kernel that doesn't include this commit.
(Side note: please don't use TCP-MD5! It's cryptographically weak. But
as long as Linux supports it, it might as well be implemented properly.)
Signed-off-by: Eric Biggers <ebiggers@kernel.org>
Link: https://patch.msgid.link/20251014215836.115616-1-ebiggers@kernel.org
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2025-10-14 14:58:36 -07:00
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2015-03-24 15:58:55 -07:00
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2006-11-14 19:07:45 -08:00
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2023-10-23 20:22:02 +01:00
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2026-04-27 10:27:26 -07:00
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2023-10-23 20:22:03 +01:00
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2023-10-23 20:22:02 +01:00
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2014-06-25 17:09:54 +03:00
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2015-09-29 07:42:49 -07:00
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2014-06-25 17:09:54 +03:00
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2020-11-30 16:36:30 +01:00
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2024-04-07 09:33:21 +00:00
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2026-03-02 20:55:27 +00:00
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2015-09-25 07:39:21 -07:00
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2014-06-25 17:09:58 +03:00
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2015-10-16 13:00:01 -07:00
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2020-08-20 12:00:52 -07:00
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2006-11-14 19:07:45 -08:00
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2020-01-09 07:59:21 -08:00
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2014-06-25 17:09:54 +03:00
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tcp: rehash onto different local ECMP path on retransmit timeout
Currently sk_rethink_txhash() re-rolls the socket's txhash on RTO, PLB,
and spurious-retransmission events, but the cached route is reused and
the new hash is not propagated into the ECMP path selection logic. Two
changes are needed to make rehash select a different local ECMP path:
1. Add __sk_dst_reset() alongside sk_rethink_txhash() in
tcp_write_timeout(), tcp_rcv_spurious_retrans(), and
tcp_plb_check_rehash() so the cached dst is invalidated and the
next transmit triggers a fresh route lookup.
2. Set fl6->mp_hash from sk_txhash (or tcp_rsk(req)->txhash for
SYN/ACK retransmits and syncookies) in tcp_v6_connect(),
inet6_sk_rebuild_header(), inet6_csk_route_req(),
inet6_csk_route_socket(), tcp_v6_send_response(), and
cookie_v6_check() so fib6_select_path() picks a path based on the
new hash.
The mp_hash override only applies to fib_multipath_hash_policy 0 (the
default L3 policy). Its hash includes the flow label, but that is 0 by
default -- np->flow_label is unset, and auto_flowlabels only computes
the on-wire label later, per packet -- so flows to the same peer share
one local path. Keying the hash on sk_txhash makes the local path
per-connection and lets a rehash re-select it. Policies 1-3 are left
unchanged.
The mp_hash assignment is factored into a small helper,
ip6_ecmp_set_mp_hash(), shared by inet6_csk_route_req(),
inet6_csk_route_socket(), tcp_v6_connect(), inet6_sk_rebuild_header(),
tcp_v6_send_response(), and cookie_v6_check(). It applies
(txhash >> 1) ?: 1 for policy 0 (the >> 1 keeps mp_hash in the 31-bit
range; ?: 1 keeps it non-zero, since 0 would fall back to
rt6_multipath_hash()). inet6_csk_route_socket() calls it only for
sk_protocol == IPPROTO_TCP so that non-TCP callers (e.g., L2TP via
inet6_csk_xmit) fall through to rt6_multipath_hash() and retain their
existing flow-key-based ECMP behavior.
tcp_v6_send_response() also sets mp_hash from the response txhash so
that a control packet (a RST from the full socket, or an ACK from a
time-wait socket) selects the same local ECMP nexthop as the
connection's txhash rather than falling back to the flow hash. The
time-wait socket's tw_txhash is copied from sk_txhash when the
connection enters TIME_WAIT, so it reflects any rehash that occurred.
Setting mp_hash explicitly is necessary because the default ECMP hash
derives from fl6->flowlabel via np->flow_label, which is not updated
from sk_txhash (REPFLOW is off by default). ip6_make_flowlabel()
cannot help either, as it runs after the route lookup.
As a consequence, for policy 0 the local ECMP path of an IPv6 TCP
flow follows sk_txhash even when fl6->flowlabel is non-zero, e.g. a
reflected (REPFLOW) or explicitly set (IPV6_FLOWLABEL_MGR) flow
label. This is intentional: only local path selection changes, so
rehash can recover from a failed path; the on-wire flow label is
unchanged.
sk_set_txhash() is moved before ip6_dst_lookup_flow() in
tcp_v6_connect() so the initial ECMP path is selected by the same
txhash that subsequent route rebuilds will use. This avoids
unintended path changes when the cached dst is naturally invalidated
(e.g., by PMTU discovery or route changes).
The rehash sites (tcp_write_timeout(), tcp_plb_check_rehash(), and
tcp_rcv_spurious_retrans()) call __sk_rethink_txhash_reset_dst(),
which re-rolls the txhash and, when it changed, drops the cached dst
so the next transmit re-runs route selection. The dst reset is
guarded by sk->sk_family == AF_INET6 since IPv4 ECMP does not
currently use sk_txhash for path selection. For IPv4-mapped IPv6
sockets this produces a redundant dst reset on a cold path
(RTO/PLB); the subsequent IPv4 route lookup returns the same result.
The helper is deliberately separate from sk_rethink_txhash() itself:
dst_negative_advice() calls sk_rethink_txhash() before its own dst op,
so resetting the dst inside sk_rethink_txhash() would skip that op
(e.g. rt6_remove_exception_rt()).
For syncookies, cookie_init_sequence() computes the cookie value
before route_req() and sets txhash so the SYN-ACK selects the same
ECMP path that cookie_v6_check() will use when the full socket is
created. cookie_tcp_reqsk_init() derives txhash from the cookie so
the full socket's ECMP path matches the SYN-ACK. Both the SYN-ACK
assignment in tcp_conn_request() and the full-socket assignment in
cookie_tcp_reqsk_init() set txhash from the cookie for IPv4 and IPv6
alike. On IPv6 this drives ECMP path selection; on IPv4, which does
not use sk_txhash for ECMP, it only affects TX-queue selection. That
selection scales the hash by its high bits (reciprocal_scale()), which
are uniform in the keyed secure_tcp_syn_cookie() output -- the MSS index
only perturbs the low bits -- so the queue distribution matches
net_tx_rndhash().
cookie_init_sequence() is split from the former version that also
called tcp_synq_overflow() and incremented SYNCOOKIESSENT; those
side effects are now in cookie_record_sent(), called after
route_req() succeeds so they are not bumped when route_req() fails.
cookie_record_sent() is guarded by CONFIG_SYN_COOKIES to
match the guard on tcp_synq_overflow(). route_req() receives 0 as
tw_isn for the syncookie path so that tcp_v6_init_req() still saves
ireq->pktopts for REPFLOW flowlabel reflection and IPv6 cmsg
options. The ecn_ok clear for syncookies without timestamps stays
after tcp_ecn_create_request() so it takes precedence.
Signed-off-by: Neil Spring <ntspring@meta.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20260615042158.1600746-2-ntspring@meta.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2026-06-14 21:21:57 -07:00
|
|
|
|
2014-06-25 17:09:54 +03:00
|
|
|
|
2015-09-29 07:42:49 -07:00
|
|
|
|
2014-06-25 17:09:54 +03:00
|
|
|
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|
tcp: rehash onto different local ECMP path on retransmit timeout
Currently sk_rethink_txhash() re-rolls the socket's txhash on RTO, PLB,
and spurious-retransmission events, but the cached route is reused and
the new hash is not propagated into the ECMP path selection logic. Two
changes are needed to make rehash select a different local ECMP path:
1. Add __sk_dst_reset() alongside sk_rethink_txhash() in
tcp_write_timeout(), tcp_rcv_spurious_retrans(), and
tcp_plb_check_rehash() so the cached dst is invalidated and the
next transmit triggers a fresh route lookup.
2. Set fl6->mp_hash from sk_txhash (or tcp_rsk(req)->txhash for
SYN/ACK retransmits and syncookies) in tcp_v6_connect(),
inet6_sk_rebuild_header(), inet6_csk_route_req(),
inet6_csk_route_socket(), tcp_v6_send_response(), and
cookie_v6_check() so fib6_select_path() picks a path based on the
new hash.
The mp_hash override only applies to fib_multipath_hash_policy 0 (the
default L3 policy). Its hash includes the flow label, but that is 0 by
default -- np->flow_label is unset, and auto_flowlabels only computes
the on-wire label later, per packet -- so flows to the same peer share
one local path. Keying the hash on sk_txhash makes the local path
per-connection and lets a rehash re-select it. Policies 1-3 are left
unchanged.
The mp_hash assignment is factored into a small helper,
ip6_ecmp_set_mp_hash(), shared by inet6_csk_route_req(),
inet6_csk_route_socket(), tcp_v6_connect(), inet6_sk_rebuild_header(),
tcp_v6_send_response(), and cookie_v6_check(). It applies
(txhash >> 1) ?: 1 for policy 0 (the >> 1 keeps mp_hash in the 31-bit
range; ?: 1 keeps it non-zero, since 0 would fall back to
rt6_multipath_hash()). inet6_csk_route_socket() calls it only for
sk_protocol == IPPROTO_TCP so that non-TCP callers (e.g., L2TP via
inet6_csk_xmit) fall through to rt6_multipath_hash() and retain their
existing flow-key-based ECMP behavior.
tcp_v6_send_response() also sets mp_hash from the response txhash so
that a control packet (a RST from the full socket, or an ACK from a
time-wait socket) selects the same local ECMP nexthop as the
connection's txhash rather than falling back to the flow hash. The
time-wait socket's tw_txhash is copied from sk_txhash when the
connection enters TIME_WAIT, so it reflects any rehash that occurred.
Setting mp_hash explicitly is necessary because the default ECMP hash
derives from fl6->flowlabel via np->flow_label, which is not updated
from sk_txhash (REPFLOW is off by default). ip6_make_flowlabel()
cannot help either, as it runs after the route lookup.
As a consequence, for policy 0 the local ECMP path of an IPv6 TCP
flow follows sk_txhash even when fl6->flowlabel is non-zero, e.g. a
reflected (REPFLOW) or explicitly set (IPV6_FLOWLABEL_MGR) flow
label. This is intentional: only local path selection changes, so
rehash can recover from a failed path; the on-wire flow label is
unchanged.
sk_set_txhash() is moved before ip6_dst_lookup_flow() in
tcp_v6_connect() so the initial ECMP path is selected by the same
txhash that subsequent route rebuilds will use. This avoids
unintended path changes when the cached dst is naturally invalidated
(e.g., by PMTU discovery or route changes).
The rehash sites (tcp_write_timeout(), tcp_plb_check_rehash(), and
tcp_rcv_spurious_retrans()) call __sk_rethink_txhash_reset_dst(),
which re-rolls the txhash and, when it changed, drops the cached dst
so the next transmit re-runs route selection. The dst reset is
guarded by sk->sk_family == AF_INET6 since IPv4 ECMP does not
currently use sk_txhash for path selection. For IPv4-mapped IPv6
sockets this produces a redundant dst reset on a cold path
(RTO/PLB); the subsequent IPv4 route lookup returns the same result.
The helper is deliberately separate from sk_rethink_txhash() itself:
dst_negative_advice() calls sk_rethink_txhash() before its own dst op,
so resetting the dst inside sk_rethink_txhash() would skip that op
(e.g. rt6_remove_exception_rt()).
For syncookies, cookie_init_sequence() computes the cookie value
before route_req() and sets txhash so the SYN-ACK selects the same
ECMP path that cookie_v6_check() will use when the full socket is
created. cookie_tcp_reqsk_init() derives txhash from the cookie so
the full socket's ECMP path matches the SYN-ACK. Both the SYN-ACK
assignment in tcp_conn_request() and the full-socket assignment in
cookie_tcp_reqsk_init() set txhash from the cookie for IPv4 and IPv6
alike. On IPv6 this drives ECMP path selection; on IPv4, which does
not use sk_txhash for ECMP, it only affects TX-queue selection. That
selection scales the hash by its high bits (reciprocal_scale()), which
are uniform in the keyed secure_tcp_syn_cookie() output -- the MSS index
only perturbs the low bits -- so the queue distribution matches
net_tx_rndhash().
cookie_init_sequence() is split from the former version that also
called tcp_synq_overflow() and incremented SYNCOOKIESSENT; those
side effects are now in cookie_record_sent(), called after
route_req() succeeds so they are not bumped when route_req() fails.
cookie_record_sent() is guarded by CONFIG_SYN_COOKIES to
match the guard on tcp_synq_overflow(). route_req() receives 0 as
tw_isn for the syncookie path so that tcp_v6_init_req() still saves
ireq->pktopts for REPFLOW flowlabel reflection and IPv6 cmsg
options. The ecn_ok clear for syncookies without timestamps stays
after tcp_ecn_create_request() so it takes precedence.
Signed-off-by: Neil Spring <ntspring@meta.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20260615042158.1600746-2-ntspring@meta.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2026-06-14 21:21:57 -07:00
|
|
|
|
2014-06-25 17:09:54 +03:00
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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|
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|
tcp: rehash onto different local ECMP path on retransmit timeout
Currently sk_rethink_txhash() re-rolls the socket's txhash on RTO, PLB,
and spurious-retransmission events, but the cached route is reused and
the new hash is not propagated into the ECMP path selection logic. Two
changes are needed to make rehash select a different local ECMP path:
1. Add __sk_dst_reset() alongside sk_rethink_txhash() in
tcp_write_timeout(), tcp_rcv_spurious_retrans(), and
tcp_plb_check_rehash() so the cached dst is invalidated and the
next transmit triggers a fresh route lookup.
2. Set fl6->mp_hash from sk_txhash (or tcp_rsk(req)->txhash for
SYN/ACK retransmits and syncookies) in tcp_v6_connect(),
inet6_sk_rebuild_header(), inet6_csk_route_req(),
inet6_csk_route_socket(), tcp_v6_send_response(), and
cookie_v6_check() so fib6_select_path() picks a path based on the
new hash.
The mp_hash override only applies to fib_multipath_hash_policy 0 (the
default L3 policy). Its hash includes the flow label, but that is 0 by
default -- np->flow_label is unset, and auto_flowlabels only computes
the on-wire label later, per packet -- so flows to the same peer share
one local path. Keying the hash on sk_txhash makes the local path
per-connection and lets a rehash re-select it. Policies 1-3 are left
unchanged.
The mp_hash assignment is factored into a small helper,
ip6_ecmp_set_mp_hash(), shared by inet6_csk_route_req(),
inet6_csk_route_socket(), tcp_v6_connect(), inet6_sk_rebuild_header(),
tcp_v6_send_response(), and cookie_v6_check(). It applies
(txhash >> 1) ?: 1 for policy 0 (the >> 1 keeps mp_hash in the 31-bit
range; ?: 1 keeps it non-zero, since 0 would fall back to
rt6_multipath_hash()). inet6_csk_route_socket() calls it only for
sk_protocol == IPPROTO_TCP so that non-TCP callers (e.g., L2TP via
inet6_csk_xmit) fall through to rt6_multipath_hash() and retain their
existing flow-key-based ECMP behavior.
tcp_v6_send_response() also sets mp_hash from the response txhash so
that a control packet (a RST from the full socket, or an ACK from a
time-wait socket) selects the same local ECMP nexthop as the
connection's txhash rather than falling back to the flow hash. The
time-wait socket's tw_txhash is copied from sk_txhash when the
connection enters TIME_WAIT, so it reflects any rehash that occurred.
Setting mp_hash explicitly is necessary because the default ECMP hash
derives from fl6->flowlabel via np->flow_label, which is not updated
from sk_txhash (REPFLOW is off by default). ip6_make_flowlabel()
cannot help either, as it runs after the route lookup.
As a consequence, for policy 0 the local ECMP path of an IPv6 TCP
flow follows sk_txhash even when fl6->flowlabel is non-zero, e.g. a
reflected (REPFLOW) or explicitly set (IPV6_FLOWLABEL_MGR) flow
label. This is intentional: only local path selection changes, so
rehash can recover from a failed path; the on-wire flow label is
unchanged.
sk_set_txhash() is moved before ip6_dst_lookup_flow() in
tcp_v6_connect() so the initial ECMP path is selected by the same
txhash that subsequent route rebuilds will use. This avoids
unintended path changes when the cached dst is naturally invalidated
(e.g., by PMTU discovery or route changes).
The rehash sites (tcp_write_timeout(), tcp_plb_check_rehash(), and
tcp_rcv_spurious_retrans()) call __sk_rethink_txhash_reset_dst(),
which re-rolls the txhash and, when it changed, drops the cached dst
so the next transmit re-runs route selection. The dst reset is
guarded by sk->sk_family == AF_INET6 since IPv4 ECMP does not
currently use sk_txhash for path selection. For IPv4-mapped IPv6
sockets this produces a redundant dst reset on a cold path
(RTO/PLB); the subsequent IPv4 route lookup returns the same result.
The helper is deliberately separate from sk_rethink_txhash() itself:
dst_negative_advice() calls sk_rethink_txhash() before its own dst op,
so resetting the dst inside sk_rethink_txhash() would skip that op
(e.g. rt6_remove_exception_rt()).
For syncookies, cookie_init_sequence() computes the cookie value
before route_req() and sets txhash so the SYN-ACK selects the same
ECMP path that cookie_v6_check() will use when the full socket is
created. cookie_tcp_reqsk_init() derives txhash from the cookie so
the full socket's ECMP path matches the SYN-ACK. Both the SYN-ACK
assignment in tcp_conn_request() and the full-socket assignment in
cookie_tcp_reqsk_init() set txhash from the cookie for IPv4 and IPv6
alike. On IPv6 this drives ECMP path selection; on IPv4, which does
not use sk_txhash for ECMP, it only affects TX-queue selection. That
selection scales the hash by its high bits (reciprocal_scale()), which
are uniform in the keyed secure_tcp_syn_cookie() output -- the MSS index
only perturbs the low bits -- so the queue distribution matches
net_tx_rndhash().
cookie_init_sequence() is split from the former version that also
called tcp_synq_overflow() and incremented SYNCOOKIESSENT; those
side effects are now in cookie_record_sent(), called after
route_req() succeeds so they are not bumped when route_req() fails.
cookie_record_sent() is guarded by CONFIG_SYN_COOKIES to
match the guard on tcp_synq_overflow(). route_req() receives 0 as
tw_isn for the syncookie path so that tcp_v6_init_req() still saves
ireq->pktopts for REPFLOW flowlabel reflection and IPv6 cmsg
options. The ecn_ok clear for syncookies without timestamps stays
after tcp_ecn_create_request() so it takes precedence.
Signed-off-by: Neil Spring <ntspring@meta.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Link: https://patch.msgid.link/20260615042158.1600746-2-ntspring@meta.com
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2026-06-14 21:21:57 -07:00
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2023-10-23 20:21:58 +01:00
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2023-10-23 20:22:00 +01:00
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2023-10-23 20:21:58 +01:00
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2023-10-23 20:22:12 +01:00
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2023-10-23 20:21:58 +01:00
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2024-06-07 00:25:55 +01:00
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2023-10-23 20:21:58 +01:00
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2024-06-07 00:25:55 +01:00
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2023-10-23 20:21:58 +01:00
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2013-09-23 11:33:32 -07:00
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2013-06-07 05:11:46 +00:00
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2013-09-23 11:33:32 -07:00
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2005-08-16 02:18:02 -03:00
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2015-10-16 21:57:46 -07:00
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2018-05-16 16:40:13 -07:00
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2018-05-16 16:40:12 -07:00
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2018-05-16 16:40:16 -07:00
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2021-01-24 13:07:14 +08:00
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2017-01-12 22:11:33 -08:00
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2015-10-16 21:57:46 -07:00
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2022-10-26 13:51:12 +00:00
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2024-10-23 16:14:51 +08:00
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2024-10-23 16:14:52 +08:00
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2024-10-23 16:14:51 +08:00
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2024-10-23 16:14:52 +08:00
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2024-10-23 16:14:51 +08:00
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2017-08-03 09:19:52 -04:00
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2017-10-05 22:21:27 -07:00
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2017-08-03 09:19:52 -04:00
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2024-09-10 15:08:22 -04:00
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2024-10-23 16:14:51 +08:00
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2024-09-10 15:08:22 -04:00
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2017-08-03 09:19:52 -04:00
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2014-10-15 14:33:21 -07:00
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2017-08-03 18:07:06 +02:00
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2014-10-15 14:33:21 -07:00
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2014-10-15 14:33:22 -07:00
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2014-10-15 14:33:21 -07:00
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2017-08-03 18:07:06 +02:00
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2014-10-15 14:33:21 -07:00
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tcp: do not pace pure ack packets
When we added pacing to TCP, we decided to let sch_fq take care
of actual pacing.
All TCP had to do was to compute sk->pacing_rate using simple formula:
sk->pacing_rate = 2 * cwnd * mss / rtt
It works well for senders (bulk flows), but not very well for receivers
or even RPC :
cwnd on the receiver can be less than 10, rtt can be around 100ms, so we
can end up pacing ACK packets, slowing down the sender.
Really, only the sender should pace, according to its own logic.
Instead of adding a new bit in skb, or call yet another flow
dissection, we tweak skb->truesize to a small value (2), and
we instruct sch_fq to use new helper and not pace pure ack.
Note this also helps TCP small queue, as ack packets present
in qdisc/NIC do not prevent sending a data packet (RPC workload)
This helps to reduce tx completion overhead, ack packets can use regular
sock_wfree() instead of tcp_wfree() which is a bit more expensive.
This has no impact in the case packets are sent to loopback interface,
as we do not coalesce ack packets (were we would detect skb->truesize
lie)
In case netem (with a delay) is used, skb_orphan_partial() also sets
skb->truesize to 1.
This patch is a combination of two patches we used for about one year at
Google.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2015-02-03 18:31:53 -08:00
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2016-03-07 14:11:05 -08:00
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2016-08-28 14:43:18 -07:00
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2016-03-14 10:52:15 -07:00
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2021-11-15 11:02:42 -08:00
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2016-03-14 10:52:15 -07:00
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2021-11-15 11:02:42 -08:00
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2016-03-14 10:52:15 -07:00
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2016-04-01 08:52:20 -07:00
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2025-08-26 12:50:27 +00:00
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2016-04-27 16:44:39 -07:00
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2016-04-01 08:52:20 -07:00
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2017-05-16 04:24:36 -07:00
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2017-06-14 11:37:14 -07:00
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2019-07-19 10:29:22 -07:00
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2020-01-11 06:12:01 +00:00
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2017-06-14 11:37:14 -07:00
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2019-08-30 12:25:48 +02:00
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tcp: ulp: diag: more info without CAP_NET_ADMIN
When introduced in commit 61723b393292 ("tcp: ulp: add functions to dump
ulp-specific information"), the whole ULP diag info has been exported
only if the requester had CAP_NET_ADMIN.
It looks like not everything is sensitive, and some info can be exported
to all users in order to ease the debugging from the userspace side
without requiring additional capabilities. Each layer should then decide
what can be exposed to everybody. The 'net_admin' boolean is then passed
to the different layers.
On kTLS side, it looks like there is nothing sensitive there: version,
cipher type, tx/rx user config type, plus some flags. So, only some
metadata about the configuration, no cryptographic info like keys, etc.
Then, everything can be exported to all users.
On MPTCP side, that's different. The MPTCP-related sequence numbers per
subflow should certainly not be exposed to everybody. For example, the
DSS mapping and ssn_offset would give all users on the system access to
narrow ranges of values for the subflow TCP sequence numbers and
MPTCP-level DSNs, and then ease packet injection. The TCP diag interface
doesn't expose the TCP sequence numbers for TCP sockets, so best to do
the same here. The rest -- token, IDs, flags -- can be exported to
everybody.
Acked-by: Mat Martineau <martineau@kernel.org>
Signed-off-by: Matthieu Baerts (NGI0) <matttbe@kernel.org>
Link: https://patch.msgid.link/20250306-net-next-tcp-ulp-diag-net-admin-v1-2-06afdd860fc9@kernel.org
Signed-off-by: Jakub Kicinski <kuba@kernel.org>
2025-03-06 12:29:28 +01:00
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2020-01-09 07:59:18 -08:00
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2017-06-14 11:37:14 -07:00
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2020-01-11 06:12:01 +00:00
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2017-06-14 11:37:14 -07:00
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tcp, ulp: add alias for all ulp modules
Lets not turn the TCP ULP lookup into an arbitrary module loader as
we only intend to load ULP modules through this mechanism, not other
unrelated kernel modules:
[root@bar]# cat foo.c
#include <sys/types.h>
#include <sys/socket.h>
#include <linux/tcp.h>
#include <linux/in.h>
int main(void)
{
int sock = socket(PF_INET, SOCK_STREAM, 0);
setsockopt(sock, IPPROTO_TCP, TCP_ULP, "sctp", sizeof("sctp"));
return 0;
}
[root@bar]# gcc foo.c -O2 -Wall
[root@bar]# lsmod | grep sctp
[root@bar]# ./a.out
[root@bar]# lsmod | grep sctp
sctp 1077248 4
libcrc32c 16384 3 nf_conntrack,nf_nat,sctp
[root@bar]#
Fix it by adding module alias to TCP ULP modules, so probing module
via request_module() will be limited to tcp-ulp-[name]. The existing
modules like kTLS will load fine given tcp-ulp-tls alias, but others
will fail to load:
[root@bar]# lsmod | grep sctp
[root@bar]# ./a.out
[root@bar]# lsmod | grep sctp
[root@bar]#
Sockmap is not affected from this since it's either built-in or not.
Fixes: 734942cc4ea6 ("tcp: ULP infrastructure")
Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
Acked-by: John Fastabend <john.fastabend@gmail.com>
Acked-by: Song Liu <songliubraving@fb.com>
Signed-off-by: Alexei Starovoitov <ast@kernel.org>
2018-08-16 21:49:06 +02:00
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2025-06-06 13:10:23 +09:00
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tcp, ulp: add alias for all ulp modules
Lets not turn the TCP ULP lookup into an arbitrary module loader as
we only intend to load ULP modules through this mechanism, not other
unrelated kernel modules:
[root@bar]# cat foo.c
#include <sys/types.h>
#include <sys/socket.h>
#include <linux/tcp.h>
#include <linux/in.h>
int main(void)
{
int sock = socket(PF_INET, SOCK_STREAM, 0);
setsockopt(sock, IPPROTO_TCP, TCP_ULP, "sctp", sizeof("sctp"));
return 0;
}
[root@bar]# gcc foo.c -O2 -Wall
[root@bar]# lsmod | grep sctp
[root@bar]# ./a.out
[root@bar]# lsmod | grep sctp
sctp 1077248 4
libcrc32c 16384 3 nf_conntrack,nf_nat,sctp
[root@bar]#
Fix it by adding module alias to TCP ULP modules, so probing module
via request_module() will be limited to tcp-ulp-[name]. The existing
modules like kTLS will load fine given tcp-ulp-tls alias, but others
will fail to load:
[root@bar]# lsmod | grep sctp
[root@bar]# ./a.out
[root@bar]# lsmod | grep sctp
[root@bar]#
Sockmap is not affected from this since it's either built-in or not.
Fixes: 734942cc4ea6 ("tcp: ULP infrastructure")
Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
Acked-by: John Fastabend <john.fastabend@gmail.com>
Acked-by: Song Liu <songliubraving@fb.com>
Signed-off-by: Alexei Starovoitov <ast@kernel.org>
2018-08-16 21:49:06 +02:00
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2021-02-23 10:49:26 -08:00
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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 02:45:58 +02:00
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2021-02-23 10:49:26 -08:00
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2021-04-06 20:21:11 -07:00
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2020-03-09 11:12:36 +00:00
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bpf: Fix wrong copied_seq calculation
'sk->copied_seq' was updated in the tcp_eat_skb() function when the action
of a BPF program was SK_REDIRECT. For other actions, like SK_PASS, the
update logic for 'sk->copied_seq' was moved to tcp_bpf_recvmsg_parser()
to ensure the accuracy of the 'fionread' feature.
It works for a single stream_verdict scenario, as it also modified
sk_data_ready->sk_psock_verdict_data_ready->tcp_read_skb
to remove updating 'sk->copied_seq'.
However, for programs where both stream_parser and stream_verdict are
active (strparser purpose), tcp_read_sock() was used instead of
tcp_read_skb() (sk_data_ready->strp_data_ready->tcp_read_sock).
tcp_read_sock() now still updates 'sk->copied_seq', leading to duplicate
updates.
In summary, for strparser + SK_PASS, copied_seq is redundantly calculated
in both tcp_read_sock() and tcp_bpf_recvmsg_parser().
The issue causes incorrect copied_seq calculations, which prevent
correct data reads from the recv() interface in user-land.
We do not want to add new proto_ops to implement a new version of
tcp_read_sock, as this would introduce code complexity [1].
We could have added noack and copied_seq to desc, and then called
ops->read_sock. However, unfortunately, other modules didn’t fully
initialize desc to zero. So, for now, we are directly calling
tcp_read_sock_noack() in tcp_bpf.c.
[1]: https://lore.kernel.org/bpf/20241218053408.437295-1-mrpre@163.com
Fixes: e5c6de5fa025 ("bpf, sockmap: Incorrectly handling copied_seq")
Suggested-by: Jakub Sitnicki <jakub@cloudflare.com>
Signed-off-by: Jiayuan Chen <mrpre@163.com>
Signed-off-by: Martin KaFai Lau <martin.lau@kernel.org>
Reviewed-by: Jakub Sitnicki <jakub@cloudflare.com>
Acked-by: John Fastabend <john.fastabend@gmail.com>
Link: https://patch.msgid.link/20250122100917.49845-3-mrpre@163.com
2025-01-22 18:09:14 +08:00
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2021-02-23 10:49:26 -08:00
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2020-03-09 11:12:36 +00:00
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2023-05-22 19:56:12 -07:00
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2022-11-29 18:40:39 +08:00
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2020-03-09 11:12:35 +00:00
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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 02:45:58 +02:00
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2021-02-23 10:49:26 -08:00
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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
2020-08-20 12:01:04 -07:00
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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>
2017-06-30 20:02:40 -07:00
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2018-01-25 16:14:09 -08:00
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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>
2017-06-30 20:02:40 -07:00
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2018-01-25 16:14:08 -08:00
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2017-12-01 10:15:04 -08:00
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bpf: Prevent unsafe access to the sock fields in the BPF timestamping callback
The subsequent patch will implement BPF TX timestamping. It will
call the sockops BPF program without holding the sock lock.
This breaks the current assumption that all sock ops programs will
hold the sock lock. The sock's fields of the uapi's bpf_sock_ops
requires this assumption.
To address this, a new "u8 is_locked_tcp_sock;" field is added. This
patch sets it in the current sock_ops callbacks. The "is_fullsock"
test is then replaced by the "is_locked_tcp_sock" test during
sock_ops_convert_ctx_access().
The new TX timestamping callbacks added in the subsequent patch will
not have this set. This will prevent unsafe access from the new
timestamping callbacks.
Potentially, we could allow read-only access. However, this would
require identifying which callback is read-safe-only and also requires
additional BPF instruction rewrites in the covert_ctx. Since the BPF
program can always read everything from a socket (e.g., by using
bpf_core_cast), this patch keeps it simple and disables all read
and write access to any socket fields through the bpf_sock_ops
UAPI from the new TX timestamping callback.
Moreover, note that some of the fields in bpf_sock_ops are specific
to tcp_sock, and sock_ops currently only supports tcp_sock. In
the future, UDP timestamping will be added, which will also break
this assumption. The same idea used in this patch will be reused.
Considering that the current sock_ops only supports tcp_sock, the
variable is named is_locked_"tcp"_sock.
Signed-off-by: Jason Xing <kerneljasonxing@gmail.com>
Signed-off-by: Martin KaFai Lau <martin.lau@kernel.org>
Link: https://patch.msgid.link/20250220072940.99994-4-kerneljasonxing@gmail.com
2025-02-20 15:29:31 +08:00
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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>
2017-06-30 20:02:40 -07:00
|
|
|
|
2017-12-01 10:15:04 -08:00
|
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|
|
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>
2017-06-30 20:02:40 -07:00
|
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|
|
2018-01-25 16:14:09 -08:00
|
|
|
|
|
|
|
|
|
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>
2017-06-30 20:02:40 -07:00
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2018-01-25 16:14:09 -08:00
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2026-06-11 09:29:18 +00:00
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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>
2017-06-30 20:02:40 -07:00
|
|
|
|
2018-01-25 16:14:09 -08:00
|
|
|
|
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>
2017-06-30 20:02:40 -07:00
|
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2018-01-25 16:14:09 -08:00
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2026-06-11 09:29:18 +00:00
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|
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>
2017-06-30 20:02:40 -07:00
|
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2017-06-30 20:02:42 -07:00
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2018-01-25 16:14:09 -08:00
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2017-06-30 20:02:42 -07:00
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2022-01-28 22:26:21 +03:00
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2017-06-30 20:02:42 -07:00
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2017-06-30 20:02:44 -07:00
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2018-01-25 16:14:09 -08:00
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2017-06-30 20:02:44 -07:00
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2017-06-30 20:02:49 -07:00
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2018-01-25 16:14:09 -08:00
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2017-06-30 20:02:49 -07:00
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2017-10-25 11:01:45 +02:00
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2024-04-26 00:17:23 +08:00
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2019-07-02 09:13:56 -07:00
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2019-07-08 14:57:21 +02:00
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2024-04-26 00:17:23 +08:00
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2019-07-02 09:13:56 -07:00
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2017-10-25 11:01:45 +02:00
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2018-04-30 10:16:10 +03:00
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2025-03-17 08:53:13 +00:00
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2018-04-30 10:16:10 +03:00
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2025-03-17 08:53:13 +00:00
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2019-05-08 16:46:14 -07:00
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2018-04-30 10:16:10 +03:00
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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>
2019-06-12 11:57:25 -07:00
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2019-06-13 21:22:35 -07:00
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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>
2019-06-12 11:57:25 -07:00
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2019-06-13 21:22:35 -07:00
|
|
|
|
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>
2019-06-12 11:57:25 -07:00
|
|
|
|
2019-06-13 21:22:35 -07:00
|
|
|
|
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>
2019-06-12 11:57:25 -07:00
|
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2023-10-23 20:21:59 +01:00
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2023-10-23 20:21:56 +01:00
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2023-10-23 20:22:13 +01:00
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2023-10-23 20:21:56 +01:00
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2023-10-23 20:22:12 +01:00
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2023-10-23 20:21:56 +01:00
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2023-10-23 20:22:13 +01:00
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2023-10-23 20:22:05 +01:00
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2023-10-23 20:21:56 +01:00
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2023-10-23 20:22:05 +01:00
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2023-10-23 20:21:56 +01:00
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2024-06-07 00:25:57 +01:00
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2023-10-23 20:22:04 +01:00
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2026-04-03 13:29:29 +02:00
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2026-04-10 17:49:50 +00:00
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2005-04-16 15:20:36 -07:00
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