- 08 2月, 2016 6 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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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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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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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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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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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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- 07 2月, 2016 1 次提交
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由 Eric Dumazet 提交于
When we acknowledge a FIN, it is not enough to ack the sequence number and queue the skb into receive queue. We also have to call tcp_fin() to properly update socket state and send proper poll() notifications. It seems we also had the problem if we received a SYN packet with the FIN flag set, but it does not seem an urgent issue, as no known implementation can do that. Fixes: 61d2bcae ("tcp: fastopen: accept data/FIN present in SYNACK message") Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Yuchung Cheng <ycheng@google.com> Cc: Neal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 06 2月, 2016 1 次提交
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由 Eric Dumazet 提交于
RFC 7413 (TCP Fast Open) 4.2.2 states that the SYNACK message MAY include data and/or FIN This patch adds support for the client side : If we receive a SYNACK with payload or FIN, queue the skb instead of ignoring it. Since we already support the same for SYN, we refactor the existing code and reuse it. Note we need to clone the skb, so this operation might fail under memory pressure. Sara Dickinson pointed out FreeBSD server Fast Open implementation was planned to generate such SYNACK in the future. The server side might be implemented on linux later. Reported-by: NSara Dickinson <sara@sinodun.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 30 1月, 2016 1 次提交
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由 Jörg Thalheim 提交于
Signed-off-by: NJörg Thalheim <joerg@higgsboson.tk> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 15 1月, 2016 2 次提交
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由 Johannes Weiner 提交于
There won't be any separate counters for socket memory consumed by protocols other than TCP in the future. Remove the indirection and link sockets directly to their owning memory cgroup. Signed-off-by: NJohannes Weiner <hannes@cmpxchg.org> Reviewed-by: NVladimir Davydov <vdavydov@virtuozzo.com> Acked-by: NDavid S. Miller <davem@davemloft.net> Signed-off-by: NAndrew Morton <akpm@linux-foundation.org> Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>
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由 Johannes Weiner 提交于
There won't be a tcp control soft limit, so integrating the memcg code into the global skmem limiting scheme complicates things unnecessarily. Replace this with simple and clear charge and uncharge calls--hidden behind a jump label--to account skb memory. Note that this is not purely aesthetic: as a result of shoehorning the per-memcg code into the same memory accounting functions that handle the global level, the old code would compare the per-memcg consumption against the smaller of the per-memcg limit and the global limit. This allowed the total consumption of multiple sockets to exceed the global limit, as long as the individual sockets stayed within bounds. After this change, the code will always compare the per-memcg consumption to the per-memcg limit, and the global consumption to the global limit, and thus close this loophole. Without a soft limit, the per-memcg memory pressure state in sockets is generally questionable. However, we did it until now, so we continue to enter it when the hard limit is hit, and packets are dropped, to let other sockets in the cgroup know that they shouldn't grow their transmit windows, either. However, keep it simple in the new callback model and leave memory pressure lazily when the next packet is accepted (as opposed to doing it synchroneously when packets are processed). When packets are dropped, network performance will already be in the toilet, so that should be a reasonable trade-off. As described above, consumption is now checked on the per-memcg level and the global level separately. Likewise, memory pressure states are maintained on both the per-memcg level and the global level, and a socket is considered under pressure when either level asserts as much. Signed-off-by: NJohannes Weiner <hannes@cmpxchg.org> Reviewed-by: NVladimir Davydov <vdavydov@virtuozzo.com> Acked-by: NDavid S. Miller <davem@davemloft.net> Signed-off-by: NAndrew Morton <akpm@linux-foundation.org> Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>
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- 11 1月, 2016 3 次提交
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由 Nikolay Borisov 提交于
This is the final part required to namespaceify the tcp keep alive mechanism. Signed-off-by: NNikolay Borisov <kernel@kyup.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Nikolay Borisov 提交于
This is required to have full tcp keepalive mechanism namespace support. Signed-off-by: NNikolay Borisov <kernel@kyup.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Nikolay Borisov 提交于
Different net namespaces might have different requirements as to the keepalive time of tcp sockets. This might be required in cases where different firewall rules are in place which require tcp timeout sockets to be increased/decreased independently of the host. 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
Avoids cluttering tcp_v4_send_reset when followup patch extends it to deal with timewait sockets. Suggested-by: NEric Dumazet <edumazet@google.com> 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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- 16 12月, 2015 1 次提交
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由 Lorenzo Colitti 提交于
This implements SOCK_DESTROY for TCP sockets. It causes all blocking calls on the socket to fail fast with ECONNABORTED and causes a protocol close of the socket. It informs the other end of the connection by sending a RST, i.e., initiating a TCP ABORT as per RFC 793. ECONNABORTED was chosen for consistency with FreeBSD. Signed-off-by: NLorenzo Colitti <lorenzo@google.com> Acked-by: NEric Dumazet <edumazet@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 3 次提交
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由 Yuchung Cheng 提交于
This patch implements the second half of RACK that uses the the most recent transmit time among all delivered packets to detect losses. tcp_rack_mark_lost() is called upon receiving a dubious ACK. It then checks if an not-yet-sacked packet was sent at least "reo_wnd" prior to the sent time of the most recently delivered. If so the packet is deemed lost. The "reo_wnd" reordering window starts with 1msec for fast loss detection and changes to min-RTT/4 when reordering is observed. We found 1msec accommodates well on tiny degree of reordering (<3 pkts) on faster links. We use min-RTT instead of SRTT because reordering is more of a path property but SRTT can be inflated by self-inflicated congestion. The factor of 4 is borrowed from the delayed early retransmit and seems to work reasonably well. Since RACK is still experimental, it is now used as a supplemental loss detection on top of existing algorithms. It is only effective after the fast recovery starts or after the timeout occurs. The fast recovery is still triggered by FACK and/or dupack threshold instead of RACK. We introduce a new sysctl net.ipv4.tcp_recovery for future experiments of loss recoveries. For now RACK can be disabled by setting it to 0. 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
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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- 19 10月, 2015 1 次提交
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由 Eric Dumazet 提交于
At the time of commit fff32699 ("tcp: reflect SYN queue_mapping into SYNACK packets") we had little ways to cope with SYN floods. We no longer need to reflect incoming skb queue mappings, and instead can pick a TX queue based on cpu cooking the SYNACK, with normal XPS affinities. Note that all SYNACK retransmits were picking TX queue 0, this no longer is a win given that SYNACK rtx are now distributed on all cpus. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 03 10月, 2015 3 次提交
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由 Eric Dumazet 提交于
If a listen backlog is very big (to avoid syncookies), then the listener sk->sk_wmem_alloc is the main source of false sharing, as we need to touch it twice per SYNACK re-transmit and TX completion. (One SYN packet takes listener lock once, but up to 6 SYNACK are generated) By attaching the skb to the request socket, we remove this source of contention. Tested: listen(fd, 10485760); // single listener (no SO_REUSEPORT) 16 RX/TX queue NIC Sustain a SYNFLOOD attack of ~320,000 SYN per second, Sending ~1,400,000 SYNACK per second. Perf profiles now show listener spinlock being next bottleneck. 20.29% [kernel] [k] queued_spin_lock_slowpath 10.06% [kernel] [k] __inet_lookup_established 5.12% [kernel] [k] reqsk_timer_handler 3.22% [kernel] [k] get_next_timer_interrupt 3.00% [kernel] [k] tcp_make_synack 2.77% [kernel] [k] ipt_do_table 2.70% [kernel] [k] run_timer_softirq 2.50% [kernel] [k] ip_finish_output 2.04% [kernel] [k] cascade 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
In this patch, we insert request sockets into TCP/DCCP regular ehash table (where ESTABLISHED and TIMEWAIT sockets are) instead of using the per listener hash table. ACK packets find SYN_RECV pseudo sockets without having to find and lock the listener. In nominal conditions, this halves pressure on listener lock. Note that this will allow for SO_REUSEPORT refinements, so that we can select a listener using cpu/numa affinities instead of the prior 'consistent hash', since only SYN packets will apply this selection logic. We will shrink listen_sock in the following patch to ease code review. Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Ying Cai <ycai@google.com> Cc: Willem de Bruijn <willemb@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
When request sockets are no longer in a per listener hash table but on regular TCP ehash, we need to access listener uid through req->rsk_listener get_openreq6() also gets a const for its request socket argument. 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 6 次提交
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由 Eric Dumazet 提交于
These functions do not change the listener socket. Goal is to make sure tcp_conn_request() is not messing with listener in a racy way. 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
Some common IPv4/IPv6 code can be factorized. Also constify cookie_init_sequence() socket argument. 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
We'll soon no longer hold listener socket lock, these functions do not modify the socket in any way. 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
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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- 29 9月, 2015 1 次提交
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由 Eric Dumazet 提交于
We found that a TCP Fast Open passive connection was vulnerable to reorders, as the exchange might look like [1] C -> S S <FO ...> <request> [2] S -> C S. ack request <options> [3] S -> C . <answer> packets [2] and [3] can be generated at almost the same time. If C receives the 3rd packet before the 2nd, it will drop it as the socket is in SYN_SENT state and expects a SYNACK. S will have to retransmit the answer. Current OOO avoidance in linux is defeated because SYNACK packets are attached to the LISTEN socket, while DATA packets are attached to the children. They might be sent by different cpus, and different TX queues might be selected. It turns out that for TFO, we created a child, which is a full blown socket in TCP_SYN_RECV state, and we simply can attach the SYNACK packet to this socket. This means that at the time tcp_sendmsg() pushes DATA packet, skb->ooo_okay will be set iff the SYNACK packet had been sent and TX completed. This removes the reorder source at the host level. We also removed the export of tcp_try_fastopen(), as it is no longer called from IPv6. 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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- 26 9月, 2015 6 次提交
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由 Eric Dumazet 提交于
This is done to make sure we do not change listener socket while sending SYNACK packets while socket lock is not held. 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
This documents fact that listener lock might not be held at the time SYNACK are sent. 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
listener socket is not locked when tcp_make_synack() is called. We better make sure no field is written. There is one exception : Since SYNACK packets are attached to the listener at this moment (or SYN_RECV child in case of Fast Open), sock_wmalloc() needs to update sk->sk_wmem_alloc, but this is done using atomic operations so this is safe. 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 TCP new listener is done, these functions will be called without socket lock being held. Make sure they don't change anything. 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
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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由 Eric Dumazet 提交于
Soon, listener socket spinlock will no longer be held, add const arguments to tcp_v[46]_init_req() to make clear these functions can not mess socket fields. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 22 9月, 2015 1 次提交
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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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- 01 9月, 2015 1 次提交
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由 Daniel Borkmann 提交于
Currently, the following case doesn't use DCTCP, even if it should: A responder has f.e. Cubic as system wide default, but for a specific route to the initiating host, DCTCP is being set in RTAX_CC_ALGO. The initiating host then uses DCTCP as congestion control, but since the initiator sets ECT(0), tcp_ecn_create_request() doesn't set ecn_ok, and we have to fall back to Reno after 3WHS completes. We were thinking on how to solve this in a minimal, non-intrusive way without bloating tcp_ecn_create_request() needlessly: lets cache the CA ecn option flag in RTAX_FEATURES. In other words, when ECT(0) is set on the SYN packet, set ecn_ok=1 iff route RTAX_FEATURES contains the unexposed (internal-only) DST_FEATURE_ECN_CA. This allows to only do a single metric feature lookup inside tcp_ecn_create_request(). Joint work with Florian Westphal. Signed-off-by: NDaniel Borkmann <daniel@iogearbox.net> Signed-off-by: NFlorian Westphal <fw@strlen.de> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 26 8月, 2015 1 次提交
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由 Eric Dumazet 提交于
When TCP pacing was added back in linux-3.12, we chose to apply a fixed ratio of 200 % against current rate, to allow probing for optimal throughput even during slow start phase, where cwnd can be doubled every other gRTT. At Google, we found it was better applying a different ratio while in Congestion Avoidance phase. This ratio was set to 120 %. We've used the normal tcp_in_slow_start() helper for a while, then tuned the condition to select the conservative ratio as soon as cwnd >= ssthresh/2 : - After cwnd reduction, it is safer to ramp up more slowly, as we approach optimal cwnd. - Initial ramp up (ssthresh == INFINITY) still allows doubling cwnd every other RTT. Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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