<feed xmlns='http://www.w3.org/2005/Atom'>
<title>linux-stable.git/net/ipv4/tcp_timer.c, branch linux-3.19.y</title>
<subtitle>Linux kernel stable tree</subtitle>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/'/>
<entry>
<title>net: Convert LIMIT_NETDEBUG to net_dbg_ratelimited</title>
<updated>2014-11-11T19:10:31+00:00</updated>
<author>
<name>Joe Perches</name>
<email>joe@perches.com</email>
</author>
<published>2014-11-11T18:59:17+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=ba7a46f16dd29f93303daeb1fee8af316c5a07f4'/>
<id>ba7a46f16dd29f93303daeb1fee8af316c5a07f4</id>
<content type='text'>
Use the more common dynamic_debug capable net_dbg_ratelimited
and remove the LIMIT_NETDEBUG macro.

All messages are still ratelimited.

Some KERN_&lt;LEVEL&gt; uses are changed to KERN_DEBUG.

This may have some negative impact on messages that were
emitted at KERN_INFO that are not not enabled at all unless
DEBUG is defined or dynamic_debug is enabled.  Even so,
these messages are now _not_ emitted by default.

This also eliminates the use of the net_msg_warn sysctl
"/proc/sys/net/core/warnings".  For backward compatibility,
the sysctl is not removed, but it has no function.  The extern
declaration of net_msg_warn is removed from sock.h and made
static in net/core/sysctl_net_core.c

Miscellanea:

o Update the sysctl documentation
o Remove the embedded uses of pr_fmt
o Coalesce format fragments
o Realign arguments

Signed-off-by: Joe Perches &lt;joe@perches.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>
Use the more common dynamic_debug capable net_dbg_ratelimited
and remove the LIMIT_NETDEBUG macro.

All messages are still ratelimited.

Some KERN_&lt;LEVEL&gt; uses are changed to KERN_DEBUG.

This may have some negative impact on messages that were
emitted at KERN_INFO that are not not enabled at all unless
DEBUG is defined or dynamic_debug is enabled.  Even so,
these messages are now _not_ emitted by default.

This also eliminates the use of the net_msg_warn sysctl
"/proc/sys/net/core/warnings".  For backward compatibility,
the sysctl is not removed, but it has no function.  The extern
declaration of net_msg_warn is removed from sock.h and made
static in net/core/sysctl_net_core.c

Miscellanea:

o Update the sysctl documentation
o Remove the embedded uses of pr_fmt
o Coalesce format fragments
o Realign arguments

Signed-off-by: Joe Perches &lt;joe@perches.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: abort orphan sockets stalling on zero window probes</title>
<updated>2014-10-01T20:27:52+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2014-09-29T20:20:38+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=b248230c34970a6c1c17c591d63b464e8d2cfc33'/>
<id>b248230c34970a6c1c17c591d63b464e8d2cfc33</id>
<content type='text'>
Currently we have two different policies for orphan sockets
that repeatedly stall on zero window ACKs. If a socket gets
a zero window ACK when it is transmitting data, the RTO is
used to probe the window. The socket is aborted after roughly
tcp_orphan_retries() retries (as in tcp_write_timeout()).

But if the socket was idle when it received the zero window ACK,
and later wants to send more data, we use the probe timer to
probe the window. If the receiver always returns zero window ACKs,
icsk_probes keeps getting reset in tcp_ack() and the orphan socket
can stall forever until the system reaches the orphan limit (as
commented in tcp_probe_timer()). This opens up a simple attack
to create lots of hanging orphan sockets to burn the memory
and the CPU, as demonstrated in the recent netdev post "TCP
connection will hang in FIN_WAIT1 after closing if zero window is
advertised." http://www.spinics.net/lists/netdev/msg296539.html

This patch follows the design in RTO-based probe: we abort an orphan
socket stalling on zero window when the probe timer reaches both
the maximum backoff and the maximum RTO. For example, an 100ms RTT
connection will timeout after roughly 153 seconds (0.3 + 0.6 +
.... + 76.8) if the receiver keeps the window shut. If the orphan
socket passes this check, but the system already has too many orphans
(as in tcp_out_of_resources()), we still abort it but we'll also
send an RST packet as the connection may still be active.

In addition, we change TCP_USER_TIMEOUT to cover (life or dead)
sockets stalled on zero-window probes. This changes the semantics
of TCP_USER_TIMEOUT slightly because it previously only applies
when the socket has pending transmission.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Reported-by: Andrey Dmitrov &lt;andrey.dmitrov@oktetlabs.ru&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 we have two different policies for orphan sockets
that repeatedly stall on zero window ACKs. If a socket gets
a zero window ACK when it is transmitting data, the RTO is
used to probe the window. The socket is aborted after roughly
tcp_orphan_retries() retries (as in tcp_write_timeout()).

But if the socket was idle when it received the zero window ACK,
and later wants to send more data, we use the probe timer to
probe the window. If the receiver always returns zero window ACKs,
icsk_probes keeps getting reset in tcp_ack() and the orphan socket
can stall forever until the system reaches the orphan limit (as
commented in tcp_probe_timer()). This opens up a simple attack
to create lots of hanging orphan sockets to burn the memory
and the CPU, as demonstrated in the recent netdev post "TCP
connection will hang in FIN_WAIT1 after closing if zero window is
advertised." http://www.spinics.net/lists/netdev/msg296539.html

This patch follows the design in RTO-based probe: we abort an orphan
socket stalling on zero window when the probe timer reaches both
the maximum backoff and the maximum RTO. For example, an 100ms RTT
connection will timeout after roughly 153 seconds (0.3 + 0.6 +
.... + 76.8) if the receiver keeps the window shut. If the orphan
socket passes this check, but the system already has too many orphans
(as in tcp_out_of_resources()), we still abort it but we'll also
send an RST packet as the connection may still be active.

In addition, we change TCP_USER_TIMEOUT to cover (life or dead)
sockets stalled on zero-window probes. This changes the semantics
of TCP_USER_TIMEOUT slightly because it previously only applies
when the socket has pending transmission.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Reported-by: Andrey Dmitrov &lt;andrey.dmitrov@oktetlabs.ru&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: avoid possible arithmetic overflows</title>
<updated>2014-09-22T20:27:10+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2014-09-22T20:19:44+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=fcdd1cf4dd63aecf86c987d7f4ec7187be5c2fbc'/>
<id>fcdd1cf4dd63aecf86c987d7f4ec7187be5c2fbc</id>
<content type='text'>
icsk_rto is a 32bit field, and icsk_backoff can reach 15 by default,
or more if some sysctl (eg tcp_retries2) are changed.

Better use 64bit to perform icsk_rto &lt;&lt; icsk_backoff operations

As Joe Perches suggested, add a helper for this.

Yuchung spotted the tcp_v4_err() case.

Signed-off-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>
icsk_rto is a 32bit field, and icsk_backoff can reach 15 by default,
or more if some sysctl (eg tcp_retries2) are changed.

Better use 64bit to perform icsk_rto &lt;&lt; icsk_backoff operations

As Joe Perches suggested, add a helper for this.

Yuchung spotted the tcp_v4_err() case.

Signed-off-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>tcp: remove TCP_SKB_CB(skb)-&gt;when</title>
<updated>2014-09-06T00:49:33+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2014-09-05T22:33:33+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=7faee5c0d514162853a343d93e4a0b6bb8bfec21'/>
<id>7faee5c0d514162853a343d93e4a0b6bb8bfec21</id>
<content type='text'>
After commit 740b0f1841f6 ("tcp: switch rtt estimations to usec resolution"),
we no longer need to maintain timestamps in two different fields.

TCP_SKB_CB(skb)-&gt;when can be removed, as same information sits in skb_mstamp.stamp_jiffies

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Acked-by: Yuchung Cheng &lt;ycheng@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>
After commit 740b0f1841f6 ("tcp: switch rtt estimations to usec resolution"),
we no longer need to maintain timestamps in two different fields.

TCP_SKB_CB(skb)-&gt;when can be removed, as same information sits in skb_mstamp.stamp_jiffies

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Acked-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: reduce spurious retransmits due to transient SACK reneging</title>
<updated>2014-08-05T23:29:33+00:00</updated>
<author>
<name>Neal Cardwell</name>
<email>ncardwell@google.com</email>
</author>
<published>2014-08-04T23:12:29+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=5ae344c949e79b8545a11db149f0a85a6e59e1f3'/>
<id>5ae344c949e79b8545a11db149f0a85a6e59e1f3</id>
<content type='text'>
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 &lt;ncardwell@google.com&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@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>
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 &lt;ncardwell@google.com&gt;
Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: snmp stats for Fast Open, SYN rtx, and data pkts</title>
<updated>2014-03-03T20:58:03+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2014-03-03T20:31:36+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=f19c29e3e391a66a273e9afebaf01917245148cd'/>
<id>f19c29e3e391a66a273e9afebaf01917245148cd</id>
<content type='text'>
Add the following snmp stats:

TCPFastOpenActiveFail: Fast Open attempts (SYN/data) failed beacuse
the remote does not accept it or the attempts timed out.

TCPSynRetrans: number of SYN and SYN/ACK retransmits to break down
retransmissions into SYN, fast-retransmits, timeout retransmits, etc.

TCPOrigDataSent: number of outgoing packets with original data (excluding
retransmission but including data-in-SYN). This counter is different from
TcpOutSegs because TcpOutSegs also tracks pure ACKs. TCPOrigDataSent is
more useful to track the TCP retransmission rate.

Change TCPFastOpenActive to track only successful Fast Opens to be symmetric to
TCPFastOpenPassive.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: Nandita Dukkipati &lt;nanditad@google.com&gt;
Signed-off-by: Lawrence Brakmo &lt;brakmo@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>
Add the following snmp stats:

TCPFastOpenActiveFail: Fast Open attempts (SYN/data) failed beacuse
the remote does not accept it or the attempts timed out.

TCPSynRetrans: number of SYN and SYN/ACK retransmits to break down
retransmissions into SYN, fast-retransmits, timeout retransmits, etc.

TCPOrigDataSent: number of outgoing packets with original data (excluding
retransmission but including data-in-SYN). This counter is different from
TcpOutSegs because TcpOutSegs also tracks pure ACKs. TCPOrigDataSent is
more useful to track the TCP retransmission rate.

Change TCPFastOpenActive to track only successful Fast Opens to be symmetric to
TCPFastOpenPassive.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: Nandita Dukkipati &lt;nanditad@google.com&gt;
Signed-off-by: Lawrence Brakmo &lt;brakmo@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: temporarily disable Fast Open on SYN timeout</title>
<updated>2013-10-30T02:50:41+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2013-10-29T17:09:05+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=c968601d174739cb1e7100c95e0eb3d2f7e91bc9'/>
<id>c968601d174739cb1e7100c95e0eb3d2f7e91bc9</id>
<content type='text'>
Fast Open currently has a fall back feature to address SYN-data being
dropped but it requires the middle-box to pass on regular SYN retry
after SYN-data. This is implemented in commit aab487435 ("net-tcp:
Fast Open client - detecting SYN-data drops")

However some NAT boxes will drop all subsequent packets after first
SYN-data and blackholes the entire connections.  An example is in
commit 356d7d8 "netfilter: nf_conntrack: fix tcp_in_window for Fast
Open".

The sender should note such incidents and fall back to use the regular
TCP handshake on subsequent attempts temporarily as well: after the
second SYN timeouts the original Fast Open SYN is most likely lost.
When such an event recurs Fast Open is disabled based on the number of
recurrences exponentially.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-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>
Fast Open currently has a fall back feature to address SYN-data being
dropped but it requires the middle-box to pass on regular SYN retry
after SYN-data. This is implemented in commit aab487435 ("net-tcp:
Fast Open client - detecting SYN-data drops")

However some NAT boxes will drop all subsequent packets after first
SYN-data and blackholes the entire connections.  An example is in
commit 356d7d8 "netfilter: nf_conntrack: fix tcp_in_window for Fast
Open".

The sender should note such incidents and fall back to use the regular
TCP handshake on subsequent attempts temporarily as well: after the
second SYN timeouts the original Fast Open SYN is most likely lost.
When such an event recurs Fast Open is disabled based on the number of
recurrences exponentially.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-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>ipv6: make lookups simpler and faster</title>
<updated>2013-10-09T04:01:25+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2013-10-03T22:42:29+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=efe4208f47f907b86f528788da711e8ab9dea44d'/>
<id>efe4208f47f907b86f528788da711e8ab9dea44d</id>
<content type='text'>
TCP listener refactoring, part 4 :

To speed up inet lookups, we moved IPv4 addresses from inet to struct
sock_common

Now is time to do the same for IPv6, because it permits us to have fast
lookups for all kind of sockets, including upcoming SYN_RECV.

Getting IPv6 addresses in TCP lookups currently requires two extra cache
lines, plus a dereference (and memory stall).

inet6_sk(sk) does the dereference of inet_sk(__sk)-&gt;pinet6

This patch is way bigger than its IPv4 counter part, because for IPv4,
we could add aliases (inet_daddr, inet_rcv_saddr), while on IPv6,
it's not doable easily.

inet6_sk(sk)-&gt;daddr becomes sk-&gt;sk_v6_daddr
inet6_sk(sk)-&gt;rcv_saddr becomes sk-&gt;sk_v6_rcv_saddr

And timewait socket also have tw-&gt;tw_v6_daddr &amp; tw-&gt;tw_v6_rcv_saddr
at the same offset.

We get rid of INET6_TW_MATCH() as INET6_MATCH() is now the generic
macro.

Signed-off-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>
TCP listener refactoring, part 4 :

To speed up inet lookups, we moved IPv4 addresses from inet to struct
sock_common

Now is time to do the same for IPv6, because it permits us to have fast
lookups for all kind of sockets, including upcoming SYN_RECV.

Getting IPv6 addresses in TCP lookups currently requires two extra cache
lines, plus a dereference (and memory stall).

inet6_sk(sk) does the dereference of inet_sk(__sk)-&gt;pinet6

This patch is way bigger than its IPv4 counter part, because for IPv4,
we could add aliases (inet_daddr, inet_rcv_saddr), while on IPv6,
it's not doable easily.

inet6_sk(sk)-&gt;daddr becomes sk-&gt;sk_v6_daddr
inet6_sk(sk)-&gt;rcv_saddr becomes sk-&gt;sk_v6_rcv_saddr

And timewait socket also have tw-&gt;tw_v6_daddr &amp; tw-&gt;tw_v6_rcv_saddr
at the same offset.

We get rid of INET6_TW_MATCH() as INET6_MATCH() is now the generic
macro.

Signed-off-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>tcp: refactor F-RTO</title>
<updated>2013-03-21T15:47:50+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2013-03-20T13:32:58+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=9b44190dc114c1720b34975b5bfc65aece112ced'/>
<id>9b44190dc114c1720b34975b5bfc65aece112ced</id>
<content type='text'>
The patch series refactor the F-RTO feature (RFC4138/5682).

This is to simplify the loss recovery processing. Existing F-RTO
was developed during the experimental stage (RFC4138) and has
many experimental features.  It takes a separate code path from
the traditional timeout processing by overloading CA_Disorder
instead of using CA_Loss state. This complicates CA_Disorder state
handling because it's also used for handling dubious ACKs and undos.
While the algorithm in the RFC does not change the congestion control,
the implementation intercepts congestion control in various places
(e.g., frto_cwnd in tcp_ack()).

The new code implements newer F-RTO RFC5682 using CA_Loss processing
path.  F-RTO becomes a small extension in the timeout processing
and interfaces with congestion control and Eifel undo modules.
It lets congestion control (module) determines how many to send
independently.  F-RTO only chooses what to send in order to detect
spurious retranmission. If timeout is found spurious it invokes
existing Eifel undo algorithms like DSACK or TCP timestamp based
detection.

The first patch removes all F-RTO code except the sysctl_tcp_frto is
left for the new implementation.  Since CA_EVENT_FRTO is removed, TCP
westwood now computes ssthresh on regular timeout CA_EVENT_LOSS event.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Acked-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>
The patch series refactor the F-RTO feature (RFC4138/5682).

This is to simplify the loss recovery processing. Existing F-RTO
was developed during the experimental stage (RFC4138) and has
many experimental features.  It takes a separate code path from
the traditional timeout processing by overloading CA_Disorder
instead of using CA_Loss state. This complicates CA_Disorder state
handling because it's also used for handling dubious ACKs and undos.
While the algorithm in the RFC does not change the congestion control,
the implementation intercepts congestion control in various places
(e.g., frto_cwnd in tcp_ack()).

The new code implements newer F-RTO RFC5682 using CA_Loss processing
path.  F-RTO becomes a small extension in the timeout processing
and interfaces with congestion control and Eifel undo modules.
It lets congestion control (module) determines how many to send
independently.  F-RTO only chooses what to send in order to detect
spurious retranmission. If timeout is found spurious it invokes
existing Eifel undo algorithms like DSACK or TCP timestamp based
detection.

The first patch removes all F-RTO code except the sysctl_tcp_frto is
left for the new implementation.  Since CA_EVENT_FRTO is removed, TCP
westwood now computes ssthresh on regular timeout CA_EVENT_LOSS event.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Acked-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>tcp: TLP loss detection.</title>
<updated>2013-03-12T12:30:34+00:00</updated>
<author>
<name>Nandita Dukkipati</name>
<email>nanditad@google.com</email>
</author>
<published>2013-03-11T10:00:44+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=9b717a8d245075ffb8e95a2dfb4ee97ce4747457'/>
<id>9b717a8d245075ffb8e95a2dfb4ee97ce4747457</id>
<content type='text'>
This is the second of the TLP patch series; it augments the basic TLP
algorithm with a loss detection scheme.

This patch implements a mechanism for loss detection when a Tail
loss probe retransmission plugs a hole thereby masking packet loss
from the sender. The loss detection algorithm relies on counting
TLP dupacks as outlined in Sec. 3 of:
http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01

The basic idea is: Sender keeps track of TLP "episode" upon
retransmission of a TLP packet. An episode ends when the sender receives
an ACK above the SND.NXT (tracked by tlp_high_seq) at the time of the
episode. We want to make sure that before the episode ends the sender
receives a "TLP dupack", indicating that the TLP retransmission was
unnecessary, so there was no loss/hole that needed plugging. If the
sender gets no TLP dupack before the end of the episode, then it reduces
ssthresh and the congestion window, because the TLP packet arriving at
the receiver probably plugged a hole.

Signed-off-by: Nandita Dukkipati &lt;nanditad@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>
This is the second of the TLP patch series; it augments the basic TLP
algorithm with a loss detection scheme.

This patch implements a mechanism for loss detection when a Tail
loss probe retransmission plugs a hole thereby masking packet loss
from the sender. The loss detection algorithm relies on counting
TLP dupacks as outlined in Sec. 3 of:
http://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-loss-probe-01

The basic idea is: Sender keeps track of TLP "episode" upon
retransmission of a TLP packet. An episode ends when the sender receives
an ACK above the SND.NXT (tracked by tlp_high_seq) at the time of the
episode. We want to make sure that before the episode ends the sender
receives a "TLP dupack", indicating that the TLP retransmission was
unnecessary, so there was no loss/hole that needed plugging. If the
sender gets no TLP dupack before the end of the episode, then it reduces
ssthresh and the congestion window, because the TLP packet arriving at
the receiver probably plugged a hole.

Signed-off-by: Nandita Dukkipati &lt;nanditad@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>
</feed>
