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- 03 12月, 2016 1 次提交
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由 Florian Westphal 提交于
jiffies based timestamps allow for easy inference of number of devices behind NAT translators and also makes tracking of hosts simpler. commit ceaa1fef ("tcp: adding a per-socket timestamp offset") added the main infrastructure that is needed for per-connection ts randomization, in particular writing/reading the on-wire tcp header format takes the offset into account so rest of stack can use normal tcp_time_stamp (jiffies). So only two items are left: - add a tsoffset for request sockets - extend the tcp isn generator to also return another 32bit number in addition to the ISN. Re-use of ISN generator also means timestamps are still monotonically increasing for same connection quadruple, i.e. PAWS will still work. Includes fixes from Eric Dumazet. Signed-off-by: NFlorian Westphal <fw@strlen.de> Acked-by: NEric Dumazet <edumazet@google.com> Acked-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 21 9月, 2016 2 次提交
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由 Soheil Hassas Yeganeh 提交于
This commit adds code to track whether the delivery rate represented by each rate_sample was limited by the application. Upon each transmit, we store in the is_app_limited field in the skb a boolean bit indicating whether there is a known "bubble in the pipe": a point in the rate sample interval where the sender was application-limited, and did not transmit even though the cwnd and pacing rate allowed it. This logic marks the flow app-limited on a write if *all* of the following are true: 1) There is less than 1 MSS of unsent data in the write queue available to transmit. 2) There is no packet in the sender's queues (e.g. in fq or the NIC tx queue). 3) The connection is not limited by cwnd. 4) There are no lost packets to retransmit. The tcp_rate_check_app_limited() code in tcp_rate.c determines whether the connection is application-limited at the moment. If the flow is application-limited, it sets the tp->app_limited field. If the flow is application-limited then that means there is effectively a "bubble" of silence in the pipe now, and this silence will be reflected in a lower bandwidth sample for any rate samples from now until we get an ACK indicating this bubble has exited the pipe: specifically, until we get an ACK for the next packet we transmit. When we send every skb we record in scb->tx.is_app_limited whether the resulting rate sample will be application-limited. The code in tcp_rate_gen() checks to see when it is safe to mark all known application-limited bubbles of silence as having exited the pipe. It does this by checking to see when the delivered count moves past the tp->app_limited marker. At this point it zeroes the tp->app_limited marker, as all known bubbles are out of the pipe. We make room for the tx.is_app_limited bit in the skb by borrowing a bit from the in_flight field used by NV to record the number of bytes in flight. The receive window in the TCP header is 16 bits, and the max receive window scaling shift factor is 14 (RFC 1323). So the max receive window offered by the TCP protocol is 2^(16+14) = 2^30. So we only need 30 bits for the tx.in_flight used by NV. Signed-off-by: NVan Jacobson <vanj@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNandita Dukkipati <nanditad@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Neal Cardwell 提交于
Refactor the TCP min_rtt code to reuse the new win_minmax library in lib/win_minmax.c to simplify the TCP code. This is a pure refactor: the functionality is exactly the same. We just moved the windowed min code to make TCP easier to read and maintain, and to allow other parts of the kernel to use the windowed min/max filter code. Signed-off-by: NVan Jacobson <vanj@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNandita Dukkipati <nanditad@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 09 9月, 2016 1 次提交
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由 Yaogong Wang 提交于
Over the years, TCP BDP has increased by several orders of magnitude, and some people are considering to reach the 2 Gbytes limit. Even with current window scale limit of 14, ~1 Gbytes maps to ~740,000 MSS. In presence of packet losses (or reorders), TCP stores incoming packets into an out of order queue, and number of skbs sitting there waiting for the missing packets to be received can be in the 10^5 range. Most packets are appended to the tail of this queue, and when packets can finally be transferred to receive queue, we scan the queue from its head. However, in presence of heavy losses, we might have to find an arbitrary point in this queue, involving a linear scan for every incoming packet, throwing away cpu caches. This patch converts it to a RB tree, to get bounded latencies. Yaogong wrote a preliminary patch about 2 years ago. Eric did the rebase, added ofo_last_skb cache, polishing and tests. Tested with network dropping between 1 and 10 % packets, with good success (about 30 % increase of throughput in stress tests) Next step would be to also use an RB tree for the write queue at sender side ;) Signed-off-by: NYaogong Wang <wygivan@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi> Acked-By: NIlpo Järvinen <ilpo.jarvinen@helsinki.fi> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 03 5月, 2016 1 次提交
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由 Eric Dumazet 提交于
We want to to make TCP stack preemptible, as draining prequeue and backlog queues can take lot of time. Many SNMP updates were assuming that BH (and preemption) was disabled. Need to convert some __NET_INC_STATS() calls to NET_INC_STATS() and some __TCP_INC_STATS() to TCP_INC_STATS() Before using this_cpu_ptr(net->ipv4.tcp_sk) in tcp_v4_send_reset() and tcp_v4_send_ack(), we add an explicit preempt disabled section. Signed-off-by: NEric Dumazet <edumazet@google.com> Acked-by: NSoheil Hassas Yeganeh <soheil@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 28 4月, 2016 2 次提交
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由 Eric Dumazet 提交于
Rename NET_INC_STATS_BH() to __NET_INC_STATS() and NET_ADD_STATS_BH() to __NET_ADD_STATS() Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
Rename TCP_INC_STATS_BH() to __TCP_INC_STATS() Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 05 4月, 2016 1 次提交
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由 Eric Dumazet 提交于
Attackers like to use SYNFLOOD targeting one 5-tuple, as they hit a single RX queue (and cpu) on the victim. If they use random sequence numbers in their SYN, we detect they do not match the expected window and send back an ACK. This patch adds a rate limitation, so that the effect of such attacks is limited to ingress only. We roughly double our ability to absorb such attacks. Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Willem de Bruijn <willemb@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Maciej Żenczykowski <maze@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 15 3月, 2016 1 次提交
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由 Martin KaFai Lau 提交于
Per RFC4898, they count segments sent/received containing a positive length data segment (that includes retransmission segments carrying data). Unlike tcpi_segs_out/in, tcpi_data_segs_out/in excludes segments carrying no data (e.g. pure ack). The patch also updates the segs_in in tcp_fastopen_add_skb() so that segs_in >= data_segs_in property is kept. Together with retransmission data, tcpi_data_segs_out gives a better signal on the rxmit rate. v6: Rebase on the latest net-next v5: Eric pointed out that checking skb->len is still needed in tcp_fastopen_add_skb() because skb can carry a FIN without data. Hence, instead of open coding segs_in and data_segs_in, tcp_segs_in() helper is used. Comment is added to the fastopen case to explain why segs_in has to be reset and tcp_segs_in() has to be called before __skb_pull(). v4: Add comment to the changes in tcp_fastopen_add_skb() and also add remark on this case in the commit message. v3: Add const modifier to the skb parameter in tcp_segs_in() v2: Rework based on recent fix by Eric: commit a9d99ce2 ("tcp: fix tcpi_segs_in after connection establishment") Signed-off-by: NMartin KaFai Lau <kafai@fb.com> Cc: Chris Rapier <rapier@psc.edu> Cc: Eric Dumazet <edumazet@google.com> Cc: Marcelo Ricardo Leitner <mleitner@redhat.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 08 3月, 2016 1 次提交
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由 Eric Dumazet 提交于
If final packet (ACK) of 3WHS is lost, it appears we do not properly account the following incoming segment into tcpi_segs_in While we are at it, starts segs_in with one, to count the SYN packet. We do not yet count number of SYN we received for a request sock, we might add this someday. packetdrill script showing proper behavior after fix : // Tests tcpi_segs_in when 3rd packet (ACK) of 3WHS is lost 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 32792 <mss 1000,sackOK,nop,nop> +0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK> +.020 < P. 1:1001(1000) ack 1 win 32792 +0 accept(3, ..., ...) = 4 +.000 %{ assert tcpi_segs_in == 2, 'tcpi_segs_in=%d' % tcpi_segs_in }% Fixes: 2efd055c ("tcp: add tcpi_segs_in and tcpi_segs_out to tcp_info") Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 08 2月, 2016 1 次提交
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由 Nikolay Borisov 提交于
Signed-off-by: NNikolay Borisov <kernel@kyup.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 23 12月, 2015 1 次提交
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由 Florian Westphal 提交于
Hannes points out that when we generate tcp reset for timewait sockets we pretend we found no socket and pass NULL sk to tcp_vX_send_reset(). Make it cope with inet tw sockets and then provide tw sk. This makes RSTs appear on correct interface when SO_BINDTODEVICE is used. Packetdrill test case: // want default route to be used, we rely on BINDTODEVICE `ip route del 192.0.2.0/24 via 192.168.0.2 dev tun0` 0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 // test case still works due to BINDTODEVICE 0.001 setsockopt(3, SOL_SOCKET, SO_BINDTODEVICE, "tun0", 4) = 0 0.100...0.200 connect(3, ..., ...) = 0 0.100 > S 0:0(0) <mss 1460,sackOK,nop,nop> 0.200 < S. 0:0(0) ack 1 win 32792 <mss 1460,sackOK,nop,nop> 0.200 > . 1:1(0) ack 1 0.210 close(3) = 0 0.210 > F. 1:1(0) ack 1 win 29200 0.300 < . 1:1(0) ack 2 win 46 // more data while in FIN_WAIT2, expect RST 1.300 < P. 1:1001(1000) ack 1 win 46 // fails without this change -- default route is used 1.301 > R 1:1(0) win 0 Reported-by: NHannes Frederic Sowa <hannes@stressinduktion.org> Signed-off-by: NFlorian Westphal <fw@strlen.de> Acked-by: NEric Dumazet <edumazet@google.com> Acked-by: NHannes Frederic Sowa <hannes@stressinduktion.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 06 11月, 2015 1 次提交
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由 Eric Dumazet 提交于
For the reasons explained in commit ce105008 ("tcp/dccp: fix ireq->pktopts race"), we need to make sure we do not access req->saved_syn unless we own the request sock. This fixes races for listeners using TCP_SAVE_SYN option. Fixes: e994b2f0 ("tcp: do not lock listener to process SYN packets") Fixes: 079096f1 ("tcp/dccp: install syn_recv requests into ehash table") Signed-off-by: NEric Dumazet <edumazet@google.com> Reported-by: NYing Cai <ycai@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 23 10月, 2015 1 次提交
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由 Eric Dumazet 提交于
Multiple cpus can process duplicates of incoming ACK messages matching a SYN_RECV request socket. This is a rare event under normal operations, but definitely can happen. Only one must win the race, otherwise corruption would occur. To fix this without adding new atomic ops, we use logic in inet_ehash_nolisten() to detect the request was present in the same ehash bucket where we try to insert the new child. If request socket was not found, we have to undo the child creation. This actually removes a spin_lock()/spin_unlock() pair in reqsk_queue_unlink() for the fast path. Fixes: e994b2f0 ("tcp: do not lock listener to process SYN packets") Fixes: 079096f1 ("tcp/dccp: install syn_recv requests into ehash table") Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 21 10月, 2015 2 次提交
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由 Yuchung Cheng 提交于
This patch is the first half of the RACK loss recovery. RACK loss recovery uses the notion of time instead of packet sequence (FACK) or counts (dupthresh). It's inspired by the previous FACK heuristic in tcp_mark_lost_retrans(): when a limited transmit (new data packet) is sacked, then current retransmitted sequence below the newly sacked sequence must been lost, since at least one round trip time has elapsed. But it has several limitations: 1) can't detect tail drops since it depends on limited transmit 2) is disabled upon reordering (assumes no reordering) 3) only enabled in fast recovery ut not timeout recovery RACK (Recently ACK) addresses these limitations with the notion of time instead: a packet P1 is lost if a later packet P2 is s/acked, as at least one round trip has passed. Since RACK cares about the time sequence instead of the data sequence of packets, it can detect tail drops when later retransmission is s/acked while FACK or dupthresh can't. For reordering RACK uses a dynamically adjusted reordering window ("reo_wnd") to reduce false positives on ever (small) degree of reordering. This patch implements tcp_advanced_rack() which tracks the most recent transmission time among the packets that have been delivered (ACKed or SACKed) in tp->rack.mstamp. This timestamp is the key to determine which packet has been lost. Consider an example that the sender sends six packets: T1: P1 (lost) T2: P2 T3: P3 T4: P4 T100: sack of P2. rack.mstamp = T2 T101: retransmit P1 T102: sack of P2,P3,P4. rack.mstamp = T4 T205: ACK of P4 since the hole is repaired. rack.mstamp = T101 We need to be careful about spurious retransmission because it may falsely advance tp->rack.mstamp by an RTT or an RTO, causing RACK to falsely mark all packets lost, just like a spurious timeout. We identify spurious retransmission by the ACK's TS echo value. If TS option is not applicable but the retransmission is acknowledged less than min-RTT ago, it is likely to be spurious. We refrain from using the transmission time of these spurious retransmissions. The second half is implemented in the next patch that marks packet lost using RACK timestamp. Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
Kathleen Nichols' algorithm for tracking the minimum RTT of a data stream over some measurement window. It uses constant space and constant time per update. Yet it almost always delivers the same minimum as an implementation that has to keep all the data in the window. The measurement window is tunable via sysctl.net.ipv4.tcp_min_rtt_wlen with a default value of 5 minutes. The algorithm keeps track of the best, 2nd best & 3rd best min values, maintaining an invariant that the measurement time of the n'th best >= n-1'th best. It also makes sure that the three values are widely separated in the time window since that bounds the worse case error when that data is monotonically increasing over the window. Upon getting a new min, we can forget everything earlier because it has no value - the new min is less than everything else in the window by definition and it's the most recent. So we restart fresh on every new min and overwrites the 2nd & 3rd choices. The same property holds for the 2nd & 3rd best. Therefore we have to maintain two invariants to maximize the information in the samples, one on values (1st.v <= 2nd.v <= 3rd.v) and the other on times (now-win <=1st.t <= 2nd.t <= 3rd.t <= now). These invariants determine the structure of the code The RTT input to the windowed filter is the minimum RTT measured from ACK or SACK, or as the last resort from TCP timestamps. The accessor tcp_min_rtt() returns the minimum RTT seen in the window. ~0U indicates it is not available. The minimum is 1usec even if the true RTT is below that. Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 13 10月, 2015 1 次提交
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由 Eric Dumazet 提交于
One 32bit hole is following skc_refcnt, use it. skc_incoming_cpu can also be an union for request_sock rcv_wnd. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 11 10月, 2015 1 次提交
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由 Eric Dumazet 提交于
Before recent TCP listener patches, we were updating listener sk->sk_rxhash before the cloning of master socket. children sk_rxhash was therefore correct after the normal 3WHS. But with lockless listener, we no longer dirty/change listener sk_rxhash as it would be racy. We need to correctly update the child sk_rxhash, otherwise first data packet wont hit correct cpu if RFS is used. Fixes: 079096f1 ("tcp/dccp: install syn_recv requests into ehash table") Signed-off-by: NEric Dumazet <edumazet@google.com> Reported-by: NWillem de Bruijn <willemb@google.com> Cc: Tom Herbert <tom@herbertland.com> Acked-by: NTom Herbert <tom@herbertland.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 03 10月, 2015 1 次提交
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由 Eric Dumazet 提交于
Once listener is lockless, its sk_state can change anytime. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 30 9月, 2015 3 次提交
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由 Eric Dumazet 提交于
This method does not touch the listener socket. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
Factorize code to get tcp header from skb. It makes no sense to duplicate code in callers. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
Once we realize tcp_rcv_synsent_state_process() does not use its 'len' argument and we get rid of it, then it becomes clear this argument is no longer used in tcp_rcv_state_process() Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 26 9月, 2015 1 次提交
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由 Eric Dumazet 提交于
Soon, listener socket wont be locked when tcp_openreq_init_rwin() is called. We need to read socket fields once, as their value could change under us. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 25 9月, 2015 1 次提交
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由 Eric Dumazet 提交于
Neal suggested to move sk_txhash init into tcp_create_openreq_child(), called both from IPv4 and IPv6. This opportunity was missed in commit 58d607d3 ("tcp: provide skb->hash to synack packets") Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 22 9月, 2015 2 次提交
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由 Eric Dumazet 提交于
When creating a timewait socket, we need to arm the timer before allowing other cpus to find it. The signal allowing cpus to find the socket is setting tw_refcnt to non zero value. As we set tw_refcnt in __inet_twsk_hashdance(), we therefore need to call inet_twsk_schedule() first. This also means we need to remove tw_refcnt changes from inet_twsk_schedule() and let the caller handle it. Note that because we use mod_timer_pinned(), we have the guarantee the timer wont expire before we set tw_refcnt as we run in BH context. To make things more readable I introduced inet_twsk_reschedule() helper. When rearming the timer, we can use mod_timer_pending() to make sure we do not rearm a canceled timer. Note: This bug can possibly trigger if packets of a flow can hit multiple cpus. This does not normally happen, unless flow steering is broken somehow. This explains this bug was spotted ~5 months after its introduction. A similar fix is needed for SYN_RECV sockets in reqsk_queue_hash_req(), but will be provided in a separate patch for proper tracking. Fixes: 789f558c ("tcp/dccp: get rid of central timewait timer") Signed-off-by: NEric Dumazet <edumazet@google.com> Reported-by: NYing Cai <ycai@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
Currently SYN/ACK RTT is measured in jiffies. For LAN the SYN/ACK RTT is often measured as 0ms or sometimes 1ms, which would affect RTT estimation and min RTT samping used by some congestion control. This patch improves SYN/ACK RTT to be usec resolution if platform supports it. While the timestamping of SYN/ACK is done in request sock, the RTT measurement is carefully arranged to avoid storing another u64 timestamp in tcp_sock. For regular handshake w/o SYNACK retransmission, the RTT is sampled right after the child socket is created and right before the request sock is released (tcp_check_req() in tcp_minisocks.c) For Fast Open the child socket is already created when SYN/ACK was sent, the RTT is sampled in tcp_rcv_state_process() after processing the final ACK an right before the request socket is released. If the SYN/ACK was retransmistted or SYN-cookie was used, we rely on TCP timestamps to measure the RTT. The sample is taken at the same place in tcp_rcv_state_process() after the timestamp values are validated in tcp_validate_incoming(). Note that we do not store TS echo value in request_sock for SYN-cookies, because the value is already stored in tp->rx_opt used by tcp_ack_update_rtt(). One side benefit is that the RTT measurement now happens before initializing congestion control (of the passive side). Therefore the congestion control can use the SYN/ACK RTT. Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 10 7月, 2015 1 次提交
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由 Eric Dumazet 提交于
inet_twsk_deschedule() calls are followed by inet_twsk_put(). Only particular case is in inet_twsk_purge() but there is no point to defer the inet_twsk_put() after re-enabling BH. Lets rename inet_twsk_deschedule() to inet_twsk_deschedule_put() and move the inet_twsk_put() inside. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 01 6月, 2015 1 次提交
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由 Neal Cardwell 提交于
Linux 3.17 and earlier are explicitly engineered so that if the app doesn't specifically request a CC module on a listener before the SYN arrives, then the child gets the system default CC when the connection is established. See tcp_init_congestion_control() in 3.17 or earlier, which says "if no choice made yet assign the current value set as default". The change ("net: tcp: assign tcp cong_ops when tcp sk is created") altered these semantics, so that children got their parent listener's congestion control even if the system default had changed after the listener was created. This commit returns to those original semantics from 3.17 and earlier, since they are the original semantics from 2007 in 4d4d3d1e ("[TCP]: Congestion control initialization."), and some Linux congestion control workflows depend on that. In summary, if a listener socket specifically sets TCP_CONGESTION to "x", or the route locks the CC module to "x", then the child gets "x". Otherwise the child gets current system default from net.ipv4.tcp_congestion_control. That's the behavior in 3.17 and earlier, and this commit restores that. Fixes: 55d8694f ("net: tcp: assign tcp cong_ops when tcp sk is created") Cc: Florian Westphal <fw@strlen.de> Cc: Daniel Borkmann <dborkman@redhat.com> Cc: Glenn Judd <glenn.judd@morganstanley.com> Cc: Stephen Hemminger <stephen@networkplumber.org> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NYuchung Cheng <ycheng@google.com> Acked-by: NDaniel Borkmann <daniel@iogearbox.net> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 22 5月, 2015 1 次提交
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由 Marcelo Ricardo Leitner 提交于
This patch tracks the total number of inbound and outbound segments on a TCP socket. One may use this number to have an idea on connection quality when compared against the retransmissions. RFC4898 named these : tcpEStatsPerfSegsIn and tcpEStatsPerfSegsOut These are a 32bit field each and can be fetched both from TCP_INFO getsockopt() if one has a handle on a TCP socket, or from inet_diag netlink facility (iproute2/ss patch will follow) Note that tp->segs_out was placed near tp->snd_nxt for good data locality and minimal performance impact, while tp->segs_in was placed near tp->bytes_received for the same reason. Join work with Eric Dumazet. Note that received SYN are accounted on the listener, but sent SYNACK are not accounted. Signed-off-by: NMarcelo Ricardo Leitner <mleitner@redhat.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 18 5月, 2015 1 次提交
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由 Florent Fourcot 提交于
commit 1d13a96c ("ipv6: tcp: fix flowlabel value in ACK messages send from TIME_WAIT") added the flow label in the last TCP packets. Unfortunately, it was not casted properly. This patch replace the buggy shift with be32_to_cpu/cpu_to_be32. Fixes: 1d13a96c ("ipv6: tcp: fix flowlabel value in ACK messages") Reported-by: NEric Dumazet <eric.dumazet@gmail.com> Signed-off-by: NFlorent Fourcot <florent.fourcot@enst-bretagne.fr> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 06 5月, 2015 1 次提交
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由 Eric Dumazet 提交于
This patch allows a server application to get the TCP SYN headers for its passive connections. This is useful if the server is doing fingerprinting of clients based on SYN packet contents. Two socket options are added: TCP_SAVE_SYN and TCP_SAVED_SYN. The first is used on a socket to enable saving the SYN headers for child connections. This can be set before or after the listen() call. The latter is used to retrieve the SYN headers for passive connections, if the parent listener has enabled TCP_SAVE_SYN. TCP_SAVED_SYN is read once, it frees the saved SYN headers. The data returned in TCP_SAVED_SYN are network (IPv4/IPv6) and TCP headers. Original patch was written by Tom Herbert, I changed it to not hold a full skb (and associated dst and conntracking reference). We have used such patch for about 3 years at Google. Signed-off-by: NEric Dumazet <edumazet@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Tested-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 24 4月, 2015 1 次提交
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由 Eric Dumazet 提交于
[ 3897.923145] BUG: unable to handle kernel NULL pointer dereference at 0000000000000080 [ 3897.931025] IP: [<ffffffffa9f27686>] reqsk_timer_handler+0x1a6/0x243 There is a race when reqsk_timer_handler() and tcp_check_req() call inet_csk_reqsk_queue_unlink() on the same req at the same time. Before commit fa76ce73 ("inet: get rid of central tcp/dccp listener timer"), listener spinlock was held and race could not happen. To solve this bug, we change reqsk_queue_unlink() to not assume req must be found, and we return a status, to conditionally release a refcount on the request sock. This also means tcp_check_req() in non fastopen case might or not consume req refcount, so tcp_v6_hnd_req() & tcp_v4_hnd_req() have to properly handle this. (Same remark for dccp_check_req() and its callers) inet_csk_reqsk_queue_drop() is now too big to be inlined, as it is called 4 times in tcp and 3 times in dccp. Fixes: fa76ce73 ("inet: get rid of central tcp/dccp listener timer") Signed-off-by: NEric Dumazet <edumazet@google.com> Reported-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 14 4月, 2015 1 次提交
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由 Eric Dumazet 提交于
Using a timer wheel for timewait sockets was nice ~15 years ago when memory was expensive and machines had a single processor. This does not scale, code is ugly and source of huge latencies (Typically 30 ms have been seen, cpus spinning on death_lock spinlock.) We can afford to use an extra 64 bytes per timewait sock and spread timewait load to all cpus to have better behavior. Tested: On following test, /proc/sys/net/ipv4/tcp_tw_recycle is set to 1 on the target (lpaa24) Before patch : lpaa23:~# ./super_netperf 200 -H lpaa24 -t TCP_CC -l 60 -- -p0,0 419594 lpaa23:~# ./super_netperf 200 -H lpaa24 -t TCP_CC -l 60 -- -p0,0 437171 While test is running, we can observe 25 or even 33 ms latencies. lpaa24:~# ping -c 1000 -i 0.02 -qn lpaa23 ... 1000 packets transmitted, 1000 received, 0% packet loss, time 20601ms rtt min/avg/max/mdev = 0.020/0.217/25.771/1.535 ms, pipe 2 lpaa24:~# ping -c 1000 -i 0.02 -qn lpaa23 ... 1000 packets transmitted, 1000 received, 0% packet loss, time 20702ms rtt min/avg/max/mdev = 0.019/0.183/33.761/1.441 ms, pipe 2 After patch : About 90% increase of throughput : lpaa23:~# ./super_netperf 200 -H lpaa24 -t TCP_CC -l 60 -- -p0,0 810442 lpaa23:~# ./super_netperf 200 -H lpaa24 -t TCP_CC -l 60 -- -p0,0 800992 And latencies are kept to minimal values during this load, even if network utilization is 90% higher : lpaa24:~# ping -c 1000 -i 0.02 -qn lpaa23 ... 1000 packets transmitted, 1000 received, 0% packet loss, time 19991ms rtt min/avg/max/mdev = 0.023/0.064/0.360/0.042 ms Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 09 4月, 2015 1 次提交
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由 Eric Dumazet 提交于
FastOpen requests are not like other regular request sockets. They do not yet use rsk_timer : tcp_fastopen_queue_check() simply manually removes one expired request from fastopenq->rskq_rst list. Therefore, tcp_check_req() must not call mod_timer_pending(), otherwise we crash because rsk_timer was not initialized. Fixes: fa76ce73 ("inet: get rid of central tcp/dccp listener timer") Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 04 4月, 2015 2 次提交
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由 Ian Morris 提交于
The ipv4 code uses a mixture of coding styles. In some instances check for non-NULL pointer is done as x != NULL and sometimes as x. x is preferred according to checkpatch and this patch makes the code consistent by adopting the latter form. No changes detected by objdiff. Signed-off-by: NIan Morris <ipm@chirality.org.uk> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Ian Morris 提交于
The ipv4 code uses a mixture of coding styles. In some instances check for NULL pointer is done as x == NULL and sometimes as !x. !x is preferred according to checkpatch and this patch makes the code consistent by adopting the latter form. No changes detected by objdiff. Signed-off-by: NIan Morris <ipm@chirality.org.uk> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 21 3月, 2015 2 次提交
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由 Eric Dumazet 提交于
One of the major issue for TCP is the SYNACK rtx handling, done by inet_csk_reqsk_queue_prune(), fired by the keepalive timer of a TCP_LISTEN socket. This function runs for awful long times, with socket lock held, meaning that other cpus needing this lock have to spin for hundred of ms. SYNACK are sent in huge bursts, likely to cause severe drops anyway. This model was OK 15 years ago when memory was very tight. We now can afford to have a timer per request sock. Timer invocations no longer need to lock the listener, and can be run from all cpus in parallel. With following patch increasing somaxconn width to 32 bits, I tested a listener with more than 4 million active request sockets, and a steady SYNFLOOD of ~200,000 SYN per second. Host was sending ~830,000 SYNACK per second. This is ~100 times more what we could achieve before this patch. Later, we will get rid of the listener hash and use ehash instead. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
When request sock are put in ehash table, the whole notion of having a previous request to update dl_next is pointless. Also, following patch will get rid of big purge timer, so we want to delete a request sock without holding listener lock. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 08 2月, 2015 2 次提交
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由 Neal Cardwell 提交于
Ensure that in state FIN_WAIT2 or TIME_WAIT, where the connection is represented by a tcp_timewait_sock, we rate limit dupacks in response to incoming packets (a) with TCP timestamps that fail PAWS checks, or (b) with sequence numbers that are out of the acceptable window. We do not send a dupack in response to out-of-window packets if it has been less than sysctl_tcp_invalid_ratelimit (default 500ms) since we last sent a dupack in response to an out-of-window packet. Reported-by: NAvery Fay <avery@mixpanel.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Neal Cardwell 提交于
Ensure that in state ESTABLISHED, where the connection is represented by a tcp_sock, we rate limit dupacks in response to incoming packets (a) with TCP timestamps that fail PAWS checks, or (b) with sequence numbers or ACK numbers that are out of the acceptable window. We do not send a dupack in response to out-of-window packets if it has been less than sysctl_tcp_invalid_ratelimit (default 500ms) since we last sent a dupack in response to an out-of-window packet. There is already a similar (although global) rate-limiting mechanism for "challenge ACKs". When deciding whether to send a challence ACK, we first consult the new per-connection rate limit, and then the global rate limit. Reported-by: NAvery Fay <avery@mixpanel.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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