<feed xmlns='http://www.w3.org/2005/Atom'>
<title>linux-stable.git/include/linux/tcp.h, branch v4.18.2</title>
<subtitle>Linux kernel stable tree</subtitle>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/'/>
<entry>
<title>tcp: add SACK compression</title>
<updated>2018-05-18T15:40:27+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2018-05-17T21:47:26+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=5d9f4262b7ea41ca9981cc790e37cca6e37c789e'/>
<id>5d9f4262b7ea41ca9981cc790e37cca6e37c789e</id>
<content type='text'>
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-&gt;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-&gt;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 &lt;edumazet@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Acked-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Toke Høiland-Jørgensen &lt;toke@toke.dk&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
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-&gt;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-&gt;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 &lt;edumazet@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Acked-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Toke Høiland-Jørgensen &lt;toke@toke.dk&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: send in-queue bytes in cmsg upon read</title>
<updated>2018-05-01T22:56:29+00:00</updated>
<author>
<name>Soheil Hassas Yeganeh</name>
<email>soheil@google.com</email>
</author>
<published>2018-05-01T19:39:15+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=b75eba76d3d72e2374fac999926dafef2997edd2'/>
<id>b75eba76d3d72e2374fac999926dafef2997edd2</id>
<content type='text'>
Applications with many concurrent connections, high variance
in receive queue length and tight memory bounds cannot
allocate worst-case buffer size to drain sockets. Knowing
the size of receive queue length, applications can optimize
how they allocate buffers to read from the socket.

The number of bytes pending on the socket is directly
available through ioctl(FIONREAD/SIOCINQ) and can be
approximated using getsockopt(MEMINFO) (rmem_alloc includes
skb overheads in addition to application data). But, both of
these options add an extra syscall per recvmsg. Moreover,
ioctl(FIONREAD/SIOCINQ) takes the socket lock.

Add the TCP_INQ socket option to TCP. When this socket
option is set, recvmsg() relays the number of bytes available
on the socket for reading to the application via the
TCP_CM_INQ control message.

Calculate the number of bytes after releasing the socket lock
to include the processed backlog, if any. To avoid an extra
branch in the hot path of recvmsg() for this new control
message, move all cmsg processing inside an existing branch for
processing receive timestamps. Since the socket lock is not held
when calculating the size of receive queue, TCP_INQ is a hint.
For example, it can overestimate the queue size by one byte,
if FIN is received.

With this method, applications can start reading from the socket
using a small buffer, and then use larger buffers based on the
remaining data when needed.

V3 change-log:
	As suggested by David Miller, added loads with barrier
	to check whether we have multiple threads calling recvmsg
	in parallel. When that happens we lock the socket to
	calculate inq.
V4 change-log:
	Removed inline from a static function.

Signed-off-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Willem de Bruijn &lt;willemb@google.com&gt;
Reviewed-by: Eric Dumazet &lt;edumazet@google.com&gt;
Reviewed-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Suggested-by: David Miller &lt;davem@davemloft.net&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Applications with many concurrent connections, high variance
in receive queue length and tight memory bounds cannot
allocate worst-case buffer size to drain sockets. Knowing
the size of receive queue length, applications can optimize
how they allocate buffers to read from the socket.

The number of bytes pending on the socket is directly
available through ioctl(FIONREAD/SIOCINQ) and can be
approximated using getsockopt(MEMINFO) (rmem_alloc includes
skb overheads in addition to application data). But, both of
these options add an extra syscall per recvmsg. Moreover,
ioctl(FIONREAD/SIOCINQ) takes the socket lock.

Add the TCP_INQ socket option to TCP. When this socket
option is set, recvmsg() relays the number of bytes available
on the socket for reading to the application via the
TCP_CM_INQ control message.

Calculate the number of bytes after releasing the socket lock
to include the processed backlog, if any. To avoid an extra
branch in the hot path of recvmsg() for this new control
message, move all cmsg processing inside an existing branch for
processing receive timestamps. Since the socket lock is not held
when calculating the size of receive queue, TCP_INQ is a hint.
For example, it can overestimate the queue size by one byte,
if FIN is received.

With this method, applications can start reading from the socket
using a small buffer, and then use larger buffers based on the
remaining data when needed.

V3 change-log:
	As suggested by David Miller, added loads with barrier
	to check whether we have multiple threads calling recvmsg
	in parallel. When that happens we lock the socket to
	calculate inq.
V4 change-log:
	Removed inline from a static function.

Signed-off-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Willem de Bruijn &lt;willemb@google.com&gt;
Reviewed-by: Eric Dumazet &lt;edumazet@google.com&gt;
Reviewed-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Suggested-by: David Miller &lt;davem@davemloft.net&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: track total bytes delivered with ECN CE marks</title>
<updated>2018-04-19T17:05:16+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2018-04-18T06:18:48+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=e21db6f69a95b846ff04e31fe0a86004cbd000d7'/>
<id>e21db6f69a95b846ff04e31fe0a86004cbd000d7</id>
<content type='text'>
Introduce a new delivered_ce stat in tcp socket to estimate
number of packets being marked with CE bits. The estimation is
done via ACKs with ECE bit. Depending on the actual receiver
behavior, the estimation could have biases.

Since the TCP sender can't really see the CE bit in the data path,
so the sender is technically counting packets marked delivered with
the "ECE / ECN-Echo" flag set.

With RFC3168 ECN, because the ECE bit is sticky, this count can
drastically overestimate the nummber of CE-marked data packets

With DCTCP-style ECN this should be reasonably precise unless there
is loss in the ACK path, in which case it's not precise.

With AccECN proposal this can be made still more precise, even in
the case some degree of ACK loss.

However this is sender's best estimate of CE information.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Reviewed-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Reviewed-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Reviewed-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Introduce a new delivered_ce stat in tcp socket to estimate
number of packets being marked with CE bits. The estimation is
done via ACKs with ECE bit. Depending on the actual receiver
behavior, the estimation could have biases.

Since the TCP sender can't really see the CE bit in the data path,
so the sender is technically counting packets marked delivered with
the "ECE / ECN-Echo" flag set.

With RFC3168 ECN, because the ECE bit is sticky, this count can
drastically overestimate the nummber of CE-marked data packets

With DCTCP-style ECN this should be reasonably precise unless there
is loss in the ACK path, in which case it's not precise.

With AccECN proposal this can be made still more precise, even in
the case some degree of ACK loss.

However this is sender's best estimate of CE information.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Reviewed-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Reviewed-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Reviewed-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>bpf: Adds field bpf_sock_ops_cb_flags to tcp_sock</title>
<updated>2018-01-26T00:41:14+00:00</updated>
<author>
<name>Lawrence Brakmo</name>
<email>brakmo@fb.com</email>
</author>
<published>2018-01-26T00:14:10+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=b13d880721729384757f235166068c315326f4a1'/>
<id>b13d880721729384757f235166068c315326f4a1</id>
<content type='text'>
Adds field bpf_sock_ops_cb_flags to tcp_sock and bpf_sock_ops. Its primary
use is to determine if there should be calls to sock_ops bpf program at
various points in the TCP code. The field is initialized to zero,
disabling the calls. A sock_ops BPF program can set it, per connection and
as necessary, when the connection is established.

It also adds support for reading and writting the field within a
sock_ops BPF program. Reading is done by accessing the field directly.
However, writing is done through the helper function
bpf_sock_ops_cb_flags_set, in order to return an error if a BPF program
is trying to set a callback that is not supported in the current kernel
(i.e. running an older kernel). The helper function returns 0 if it was
able to set all of the bits set in the argument, a positive number
containing the bits that could not be set, or -EINVAL if the socket is
not a full TCP socket.

Examples of where one could call the bpf program:

1) When RTO fires
2) When a packet is retransmitted
3) When the connection terminates
4) When a packet is sent
5) When a packet is received

Signed-off-by: Lawrence Brakmo &lt;brakmo@fb.com&gt;
Acked-by: Alexei Starovoitov &lt;ast@kernel.org&gt;
Signed-off-by: Alexei Starovoitov &lt;ast@kernel.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Adds field bpf_sock_ops_cb_flags to tcp_sock and bpf_sock_ops. Its primary
use is to determine if there should be calls to sock_ops bpf program at
various points in the TCP code. The field is initialized to zero,
disabling the calls. A sock_ops BPF program can set it, per connection and
as necessary, when the connection is established.

It also adds support for reading and writting the field within a
sock_ops BPF program. Reading is done by accessing the field directly.
However, writing is done through the helper function
bpf_sock_ops_cb_flags_set, in order to return an error if a BPF program
is trying to set a callback that is not supported in the current kernel
(i.e. running an older kernel). The helper function returns 0 if it was
able to set all of the bits set in the argument, a positive number
containing the bits that could not be set, or -EINVAL if the socket is
not a full TCP socket.

Examples of where one could call the bpf program:

1) When RTO fires
2) When a packet is retransmitted
3) When the connection terminates
4) When a packet is sent
5) When a packet is received

Signed-off-by: Lawrence Brakmo &lt;brakmo@fb.com&gt;
Acked-by: Alexei Starovoitov &lt;ast@kernel.org&gt;
Signed-off-by: Alexei Starovoitov &lt;ast@kernel.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: avoid integer overflows in tcp_rcv_space_adjust()</title>
<updated>2017-12-12T15:53:04+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2017-12-11T01:55:03+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=607065bad9931e72207b0cac365d7d4abc06bd99'/>
<id>607065bad9931e72207b0cac365d7d4abc06bd99</id>
<content type='text'>
When using large tcp_rmem[2] values (I did tests with 500 MB),
I noticed overflows while computing rcvwin.

Lets fix this before the following patch.

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Acked-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Acked-by: Wei Wang &lt;weiwan@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
When using large tcp_rmem[2] values (I did tests with 500 MB),
I noticed overflows while computing rcvwin.

Lets fix this before the following patch.

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Acked-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Acked-by: Wei Wang &lt;weiwan@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: invalidate rate samples during SACK reneging</title>
<updated>2017-12-08T15:07:02+00:00</updated>
<author>
<name>Yousuk Seung</name>
<email>ysseung@google.com</email>
</author>
<published>2017-12-07T21:41:34+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=d4761754b4fb2ef8d9a1e9d121c4bec84e1fe292'/>
<id>d4761754b4fb2ef8d9a1e9d121c4bec84e1fe292</id>
<content type='text'>
Mark tcp_sock during a SACK reneging event and invalidate rate samples
while marked. Such rate samples may overestimate bw by including packets
that were SACKed before reneging.

&lt; ack 6001 win 10000 sack 7001:38001
&lt; ack 7001 win 0 sack 8001:38001 // Reneg detected
&gt; seq 7001:8001 // RTO, SACK cleared.
&lt; ack 38001 win 10000

In above example the rate sample taken after the last ack will count
7001-38001 as delivered while the actual delivery rate likely could
be much lower i.e. 7001-8001.

This patch adds a new field tcp_sock.sack_reneg and marks it when we
declare SACK reneging and entering TCP_CA_Loss, and unmarks it after
the last rate sample was taken before moving back to TCP_CA_Open. This
patch also invalidates rate samples taken while tcp_sock.is_sack_reneg
is set.

Fixes: b9f64820fb22 ("tcp: track data delivery rate for a TCP connection")
Signed-off-by: Yousuk Seung &lt;ysseung@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Acked-by: Eric Dumazet &lt;edumazet@google.com&gt;
Acked-by: Priyaranjan Jha &lt;priyarjha@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Mark tcp_sock during a SACK reneging event and invalidate rate samples
while marked. Such rate samples may overestimate bw by including packets
that were SACKed before reneging.

&lt; ack 6001 win 10000 sack 7001:38001
&lt; ack 7001 win 0 sack 8001:38001 // Reneg detected
&gt; seq 7001:8001 // RTO, SACK cleared.
&lt; ack 38001 win 10000

In above example the rate sample taken after the last ack will count
7001-38001 as delivered while the actual delivery rate likely could
be much lower i.e. 7001-8001.

This patch adds a new field tcp_sock.sack_reneg and marks it when we
declare SACK reneging and entering TCP_CA_Loss, and unmarks it after
the last rate sample was taken before moving back to TCP_CA_Open. This
patch also invalidates rate samples taken while tcp_sock.is_sack_reneg
is set.

Fixes: b9f64820fb22 ("tcp: track data delivery rate for a TCP connection")
Signed-off-by: Yousuk Seung &lt;ysseung@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Acked-by: Eric Dumazet &lt;edumazet@google.com&gt;
Acked-by: Priyaranjan Jha &lt;priyarjha@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: use sequence distance to detect reordering</title>
<updated>2017-11-11T09:53:16+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2017-11-08T21:01:27+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=737ff314563ca27f044f9a3a041e9d42491ef7ce'/>
<id>737ff314563ca27f044f9a3a041e9d42491ef7ce</id>
<content type='text'>
Replace the reordering distance measurement in packet unit with
sequence based approach. Previously it trackes the number of "packets"
toward the forward ACK (i.e.  highest sacked sequence)in a state
variable "fackets_out".

Precisely measuring reordering degree on packet distance has not much
benefit, as the degree constantly changes by factors like path, load,
and congestion window. It is also complicated and prone to arcane bugs.
This patch replaces with sequence-based approach that's much simpler.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Reviewed-by: Eric Dumazet &lt;edumazet@google.com&gt;
Reviewed-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Reviewed-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Reviewed-by: Priyaranjan Jha &lt;priyarjha@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Replace the reordering distance measurement in packet unit with
sequence based approach. Previously it trackes the number of "packets"
toward the forward ACK (i.e.  highest sacked sequence)in a state
variable "fackets_out".

Precisely measuring reordering degree on packet distance has not much
benefit, as the degree constantly changes by factors like path, load,
and congestion window. It is also complicated and prone to arcane bugs.
This patch replaces with sequence-based approach that's much simpler.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Reviewed-by: Eric Dumazet &lt;edumazet@google.com&gt;
Reviewed-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Reviewed-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Reviewed-by: Priyaranjan Jha &lt;priyarjha@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: retire FACK loss detection</title>
<updated>2017-11-11T09:53:16+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2017-11-08T21:01:26+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=713bafea92920103cd3d361657406cf04d0e22dd'/>
<id>713bafea92920103cd3d361657406cf04d0e22dd</id>
<content type='text'>
FACK loss detection has been disabled by default and the
successor RACK subsumed FACK and can handle reordering better.
This patch removes FACK to simplify TCP loss recovery.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Reviewed-by: Eric Dumazet &lt;edumazet@google.com&gt;
Reviewed-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Reviewed-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Reviewed-by: Priyaranjan Jha &lt;priyarjha@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
FACK loss detection has been disabled by default and the
successor RACK subsumed FACK and can handle reordering better.
This patch removes FACK to simplify TCP loss recovery.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Reviewed-by: Eric Dumazet &lt;edumazet@google.com&gt;
Reviewed-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Reviewed-by: Soheil Hassas Yeganeh &lt;soheil@google.com&gt;
Reviewed-by: Priyaranjan Jha &lt;priyarjha@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: higher throughput under reordering with adaptive RACK reordering wnd</title>
<updated>2017-11-05T14:15:42+00:00</updated>
<author>
<name>Priyaranjan Jha</name>
<email>priyarjha@google.com</email>
</author>
<published>2017-11-03T23:38:48+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=1f2556916d974cfb62b6af51660186b5f58bd869'/>
<id>1f2556916d974cfb62b6af51660186b5f58bd869</id>
<content type='text'>
Currently TCP RACK loss detection does not work well if packets are
being reordered beyond its static reordering window (min_rtt/4).Under
such reordering it may falsely trigger loss recoveries and reduce TCP
throughput significantly.

This patch improves that by increasing and reducing the reordering
window based on DSACK, which is now supported in major TCP implementations.
It makes RACK's reo_wnd adaptive based on DSACK and no. of recoveries.

- If DSACK is received, increment reo_wnd by min_rtt/4 (upper bounded
  by srtt), since there is possibility that spurious retransmission was
  due to reordering delay longer than reo_wnd.

- Persist the current reo_wnd value for TCP_RACK_RECOVERY_THRESH (16)
  no. of successful recoveries (accounts for full DSACK-based loss
  recovery undo). After that, reset it to default (min_rtt/4).

- At max, reo_wnd is incremented only once per rtt. So that the new
  DSACK on which we are reacting, is due to the spurious retx (approx)
  after the reo_wnd has been updated last time.

- reo_wnd is tracked in terms of steps (of min_rtt/4), rather than
  absolute value to account for change in rtt.

In our internal testing, we observed significant increase in throughput,
in scenarios where reordering exceeds min_rtt/4 (previous static value).

Signed-off-by: Priyaranjan Jha &lt;priyarjha@google.com&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Currently TCP RACK loss detection does not work well if packets are
being reordered beyond its static reordering window (min_rtt/4).Under
such reordering it may falsely trigger loss recoveries and reduce TCP
throughput significantly.

This patch improves that by increasing and reducing the reordering
window based on DSACK, which is now supported in major TCP implementations.
It makes RACK's reo_wnd adaptive based on DSACK and no. of recoveries.

- If DSACK is received, increment reo_wnd by min_rtt/4 (upper bounded
  by srtt), since there is possibility that spurious retransmission was
  due to reordering delay longer than reo_wnd.

- Persist the current reo_wnd value for TCP_RACK_RECOVERY_THRESH (16)
  no. of successful recoveries (accounts for full DSACK-based loss
  recovery undo). After that, reset it to default (min_rtt/4).

- At max, reo_wnd is incremented only once per rtt. So that the new
  DSACK on which we are reacting, is due to the spurious retx (approx)
  after the reo_wnd has been updated last time.

- reo_wnd is tracked in terms of steps (of min_rtt/4), rather than
  absolute value to account for change in rtt.

In our internal testing, we observed significant increase in throughput,
in scenarios where reordering exceeds min_rtt/4 (previous static value).

Signed-off-by: Priyaranjan Jha &lt;priyarjha@google.com&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: TCP experimental option for SMC</title>
<updated>2017-10-26T09:00:29+00:00</updated>
<author>
<name>Ursula Braun</name>
<email>ubraun@linux.vnet.ibm.com</email>
</author>
<published>2017-10-25T09:01:45+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=60e2a7780793bae0debc275a9ccd57f7da0cf195'/>
<id>60e2a7780793bae0debc275a9ccd57f7da0cf195</id>
<content type='text'>
The SMC protocol [1] relies on the use of a new TCP experimental
option [2, 3]. With this option, SMC capabilities are exchanged
between peers during the TCP three way handshake. This patch adds
support for this experimental option to TCP.

References:
[1] SMC-R Informational RFC: http://www.rfc-editor.org/info/rfc7609
[2] Shared Use of TCP Experimental Options RFC 6994:
    https://tools.ietf.org/rfc/rfc6994.txt
[3] IANA ExID SMCR:
http://www.iana.org/assignments/tcp-parameters/tcp-parameters.xhtml#tcp-exids

Signed-off-by: Ursula Braun &lt;ubraun@linux.vnet.ibm.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
The SMC protocol [1] relies on the use of a new TCP experimental
option [2, 3]. With this option, SMC capabilities are exchanged
between peers during the TCP three way handshake. This patch adds
support for this experimental option to TCP.

References:
[1] SMC-R Informational RFC: http://www.rfc-editor.org/info/rfc7609
[2] Shared Use of TCP Experimental Options RFC 6994:
    https://tools.ietf.org/rfc/rfc6994.txt
[3] IANA ExID SMCR:
http://www.iana.org/assignments/tcp-parameters/tcp-parameters.xhtml#tcp-exids

Signed-off-by: Ursula Braun &lt;ubraun@linux.vnet.ibm.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
</feed>
