- 03 5月, 2016 2 次提交
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由 Eric Dumazet 提交于
AFAIK, nothing in current TCP stack absolutely wants BH being disabled once socket is owned by a thread running in process context. As mentioned in my prior patch ("tcp: give prequeue mode some care"), processing a batch of packets might take time, better not block BH at all. Signed-off-by: NEric Dumazet <edumazet@google.com> Acked-by: NSoheil Hassas Yeganeh <soheil@google.com> Acked-by: NAlexei Starovoitov <ast@kernel.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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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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- 29 4月, 2016 2 次提交
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由 Martin KaFai Lau 提交于
This patch: 1. Prevent next_skb from coalescing to the prev_skb if TCP_SKB_CB(prev_skb)->eor is set 2. Update the TCP_SKB_CB(prev_skb)->eor if coalescing is allowed Packetdrill script for testing: ~~~~~~ +0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10` +0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1` +0 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7> 0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7> 0.200 < . 1:1(0) ack 1 win 257 0.200 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0 0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730 0.200 sendto(4, ..., 730, MSG_EOR, ..., ...) = 730 0.200 write(4, ..., 11680) = 11680 0.200 > P. 1:731(730) ack 1 0.200 > P. 731:1461(730) ack 1 0.200 > . 1461:8761(7300) ack 1 0.200 > P. 8761:13141(4380) ack 1 0.300 < . 1:1(0) ack 1 win 257 <sack 1461:13141,nop,nop> 0.300 > P. 1:731(730) ack 1 0.300 > P. 731:1461(730) ack 1 0.400 < . 1:1(0) ack 13141 win 257 0.400 close(4) = 0 0.400 > F. 13141:13141(0) ack 1 0.500 < F. 1:1(0) ack 13142 win 257 0.500 > . 13142:13142(0) ack 2 Signed-off-by: NMartin KaFai Lau <kafai@fb.com> Cc: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Soheil Hassas Yeganeh <soheil@google.com> Cc: Willem de Bruijn <willemb@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: 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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由 Soheil Hassas Yeganeh 提交于
The SKBTX_ACK_TSTAMP flag is set in skb_shinfo->tx_flags when the timestamp of the TCP acknowledgement should be reported on error queue. Since accessing skb_shinfo is likely to incur a cache-line miss at the time of receiving the ack, the txstamp_ack bit was added in tcp_skb_cb, which is set iff the SKBTX_ACK_TSTAMP flag is set for an skb. This makes SKBTX_ACK_TSTAMP flag redundant. Remove the SKBTX_ACK_TSTAMP and instead use the txstamp_ack bit everywhere. Note that this frees one bit in shinfo->tx_flags. Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com> Acked-by: NMartin KaFai Lau <kafai@fb.com> Suggested-by: NWillem de Bruijn <willemb@google.com> Acked-by: NEric Dumazet <edumazet@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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- 26 4月, 2016 1 次提交
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由 Eric Dumazet 提交于
We now have proper per-listener but also per network namespace counters for SYN packets that might be dropped. We replace the kfree_skb() by consume_skb() to be drop monitor [1] friendly, and remove an obsolete comment. FastOpen SYN packets can carry payload in them just fine. [1] perf record -a -g -e skb:kfree_skb sleep 1; perf report Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 25 4月, 2016 1 次提交
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由 Eric Dumazet 提交于
Linux TCP stack painfully segments all TSO/GSO packets before retransmits. This was fine back in the days when TSO/GSO were emerging, with their bugs, but we believe the dark age is over. Keeping big packets in write queues, but also in stack traversal has a lot of benefits. - Less memory overhead, because write queues have less skbs - Less cpu overhead at ACK processing. - Better SACK processing, as lot of studies mentioned how awful linux was at this ;) - Less cpu overhead to send the rtx packets (IP stack traversal, netfilter traversal, drivers...) - Better latencies in presence of losses. - Smaller spikes in fq like packet schedulers, as retransmits are not constrained by TCP Small Queues. 1 % packet losses are common today, and at 100Gbit speeds, this translates to ~80,000 losses per second. Losses are often correlated, and we see many retransmit events leading to 1-MSS train of packets, at the time hosts are already under stress. Signed-off-by: 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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- 22 4月, 2016 2 次提交
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由 Martin KaFai Lau 提交于
After receiving sacks, tcp_shifted_skb() will collapse skbs if possible. tx_flags and tskey also have to be merged. This patch reuses the tcp_skb_collapse_tstamp() to handle them. BPF Output Before: ~~~~~ <no-output-due-to-missing-tstamp-event> BPF Output After: ~~~~~ <...>-2024 [007] d.s. 88.644374: : ee_data:14599 Packetdrill Script: ~~~~~ +0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10` +0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1` +0 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7> 0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7> 0.200 < . 1:1(0) ack 1 win 257 0.200 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0 0.200 write(4, ..., 1460) = 1460 +0 setsockopt(4, SOL_SOCKET, 37, [2688], 4) = 0 0.200 write(4, ..., 13140) = 13140 0.200 > P. 1:1461(1460) ack 1 0.200 > . 1461:8761(7300) ack 1 0.200 > P. 8761:14601(5840) ack 1 0.300 < . 1:1(0) ack 1 win 257 <sack 1461:14601,nop,nop> 0.300 > P. 1:1461(1460) ack 1 0.400 < . 1:1(0) ack 14601 win 257 0.400 close(4) = 0 0.400 > F. 14601:14601(0) ack 1 0.500 < F. 1:1(0) ack 14602 win 257 0.500 > . 14602:14602(0) ack 2 Signed-off-by: NMartin KaFai Lau <kafai@fb.com> Cc: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Soheil Hassas Yeganeh <soheil@google.com> Cc: Willem de Bruijn <willemb@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: NSoheil Hassas Yeganeh <soheil@google.com> Tested-by: NSoheil Hassas Yeganeh <soheil@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Martin KaFai Lau 提交于
Assuming SOF_TIMESTAMPING_TX_ACK is on. When dup acks are received, it could incorrectly think that a skb has already been acked and queue a SCM_TSTAMP_ACK cmsg to the sk->sk_error_queue. In tcp_ack_tstamp(), it checks 'between(shinfo->tskey, prior_snd_una, tcp_sk(sk)->snd_una - 1)'. If prior_snd_una == tcp_sk(sk)->snd_una like the following packetdrill script, between() returns true but the tskey is actually not acked. e.g. try between(3, 2, 1). The fix is to replace between() with one before() and one !before(). By doing this, the -1 offset on the tcp_sk(sk)->snd_una can also be removed. A packetdrill script is used to reproduce the dup ack scenario. Due to the lacking cmsg support in packetdrill (may be I cannot find it), a BPF prog is used to kprobe to sock_queue_err_skb() and print out the value of serr->ee.ee_data. Both the packetdrill and the bcc BPF script is attached at the end of this commit message. BPF Output Before Fix: ~~~~~~ <...>-2056 [001] d.s. 433.927987: : ee_data:1459 #incorrect packetdrill-2056 [001] d.s. 433.929563: : ee_data:1459 #incorrect packetdrill-2056 [001] d.s. 433.930765: : ee_data:1459 #incorrect packetdrill-2056 [001] d.s. 434.028177: : ee_data:1459 packetdrill-2056 [001] d.s. 434.029686: : ee_data:14599 BPF Output After Fix: ~~~~~~ <...>-2049 [000] d.s. 113.517039: : ee_data:1459 <...>-2049 [000] d.s. 113.517253: : ee_data:14599 BCC BPF Script: ~~~~~~ #!/usr/bin/env python from __future__ import print_function from bcc import BPF bpf_text = """ #include <uapi/linux/ptrace.h> #include <net/sock.h> #include <bcc/proto.h> #include <linux/errqueue.h> #ifdef memset #undef memset #endif int trace_err_skb(struct pt_regs *ctx) { struct sk_buff *skb = (struct sk_buff *)ctx->si; struct sock *sk = (struct sock *)ctx->di; struct sock_exterr_skb *serr; u32 ee_data = 0; if (!sk || !skb) return 0; serr = SKB_EXT_ERR(skb); bpf_probe_read(&ee_data, sizeof(ee_data), &serr->ee.ee_data); bpf_trace_printk("ee_data:%u\\n", ee_data); return 0; }; """ b = BPF(text=bpf_text) b.attach_kprobe(event="sock_queue_err_skb", fn_name="trace_err_skb") print("Attached to kprobe") b.trace_print() Packetdrill Script: ~~~~~~ +0 `sysctl -q -w net.ipv4.tcp_min_tso_segs=10` +0 `sysctl -q -w net.ipv4.tcp_no_metrics_save=1` +0 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 0.100 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7> 0.100 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 7> 0.200 < . 1:1(0) ack 1 win 257 0.200 accept(3, ..., ...) = 4 +0 setsockopt(4, SOL_TCP, TCP_NODELAY, [1], 4) = 0 +0 setsockopt(4, SOL_SOCKET, 37, [2688], 4) = 0 0.200 write(4, ..., 1460) = 1460 0.200 write(4, ..., 13140) = 13140 0.200 > P. 1:1461(1460) ack 1 0.200 > . 1461:8761(7300) ack 1 0.200 > P. 8761:14601(5840) ack 1 0.300 < . 1:1(0) ack 1 win 257 <sack 1461:2921,nop,nop> 0.300 < . 1:1(0) ack 1 win 257 <sack 1461:4381,nop,nop> 0.300 < . 1:1(0) ack 1 win 257 <sack 1461:5841,nop,nop> 0.300 > P. 1:1461(1460) ack 1 0.400 < . 1:1(0) ack 14601 win 257 0.400 close(4) = 0 0.400 > F. 14601:14601(0) ack 1 0.500 < F. 1:1(0) ack 14602 win 257 0.500 > . 14602:14602(0) ack 2 Signed-off-by: NMartin KaFai Lau <kafai@fb.com> Cc: Eric Dumazet <edumazet@google.com> Cc: Neal Cardwell <ncardwell@google.com> Cc: Soheil Hassas Yeganeh <soheil.kdev@gmail.com> Cc: Willem de Bruijn <willemb@google.com> Cc: Yuchung Cheng <ycheng@google.com> Acked-by: NSoheil Hassas Yeganeh <soheil@google.com> Tested-by: NSoheil Hassas Yeganeh <soheil@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 16 4月, 2016 2 次提交
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由 Eric Dumazet 提交于
Last known hot point during SYNFLOOD attack is the clearing of rx_opt.saw_tstamp in tcp_rcv_state_process() It is not needed for a listener, so we move it where it matters. Performance while a SYNFLOOD hits a single listener socket went from 5 Mpps to 6 Mpps on my test server (24 cores, 8 NIC RX queues) 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 removing sk_refcnt manipulation on synflood, I missed that using skb_set_owner_w() was racy, if sk->sk_wmem_alloc had already transitioned to 0. We should hold sk_refcnt instead, but this is a big deal under attack. (Doing so increase performance from 3.2 Mpps to 3.8 Mpps only) In this patch, I chose to not attach a socket to syncookies skb. Performance is now 5 Mpps instead of 3.2 Mpps. Following patch will remove last known false sharing in tcp_rcv_state_process() Fixes: 3b24d854 ("tcp/dccp: do not touch listener sk_refcnt under synflood") 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 3 次提交
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由 Eric Dumazet 提交于
Goal: packets dropped by a listener are accounted for. This adds tcp_listendrop() helper, and clears sk_drops in sk_clone_lock() so that children do not inherit their parent drop count. Note that we no longer increment LINUX_MIB_LISTENDROPS counter when sending a SYNCOOKIE, since the SYN packet generated a SYNACK. We already have a separate LINUX_MIB_SYNCOOKIESSENT 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
Now ss can report sk_drops, we can instruct TCP to increment this per socket counter when it drops an incoming frame, to refine monitoring and debugging. Following patch takes care of listeners drops. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Soheil Hassas Yeganeh 提交于
Currently, to avoid a cache line miss for accessing skb_shinfo, tcp_ack_tstamp skips socket that do not have SOF_TIMESTAMPING_TX_ACK bit set in sk_tsflags. This is implemented based on an implicit assumption that the SOF_TIMESTAMPING_TX_ACK is set via socket options for the duration that ACK timestamps are needed. To implement per-write timestamps, this check should be removed and replaced with a per-packet alternative that quickly skips packets missing ACK timestamps marks without a cache-line miss. To enable per-packet marking without a cache line miss, use one bit in TCP_SKB_CB to mark a whether a SKB might need a ack tx timestamp or not. Further checks in tcp_ack_tstamp are not modified and work as before. Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com> Acked-by: NWillem de Bruijn <willemb@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 03 4月, 2016 1 次提交
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由 Yuchung Cheng 提交于
For non-SACK connections, cwnd is lowered to inflight plus 3 packets when the recovery ends. This is an optional feature in the NewReno RFC 2582 to reduce the potential burst when cwnd is "re-opened" after recovery and inflight is low. This feature is questionably effective because of PRR: when the recovery ends (i.e., snd_una == high_seq) NewReno holds the CA_Recovery state for another round trip to prevent false fast retransmits. But if the inflight is low, PRR will overwrite the moderated cwnd in tcp_cwnd_reduction() later regardlessly. So if a receiver responds bogus ACKs (i.e., acking future data) to speed up transfer after recovery, it can only induce a burst up to a window worth of data packets by acking up to SND.NXT. A restart from (short) idle or receiving streched ACKs can both cause such bursts as well. On the other hand, if the recovery ends because the sender detects the losses were spurious (e.g., reordering). This feature unconditionally lowers a reverted cwnd even though nothing was lost. By principle loss recovery module should not update cwnd. Further pacing is much more effective to reduce burst. Hence this patch removes the cwnd moderation feature. v2 changes: revised commit message on bogus ACKs and burst, and missing signature Signed-off-by: NMatt Mathis <mattmathis@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com> Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 17 2月, 2016 1 次提交
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由 Eric Dumazet 提交于
There are some cases where rtt_us derives from deltas of jiffies, instead of using usec timestamps. Since we want to track minimal rtt, better to assume a delta of 0 jiffie might be in fact be very close to 1 jiffie. It is kind of sad jiffies_to_usecs(1) calls a function instead of simply using a constant. Fixes: f6722583 ("tcp: track min RTT using windowed min-filter") Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Cc: Yuchung Cheng <ycheng@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 08 2月, 2016 8 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 Yuchung Cheng 提交于
Refactor and consolidate cwnd and rate updates into a new function tcp_cong_control(). Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
This change enables congestion control to update cwnd based on not only packet cumulatively acked but also packets delivered out-of-order. This makes congestion control robust against packet reordering because it may raise cwnd as long as packets are being delivered once reordering has been detected (i.e., it only cares the amount of packets delivered, not the ordering among them). Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
A small refactoring that gets number of packets cumulatively acked from tcp_clean_rtx_queue() directly. Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
This patch changes the accounting of how many packets are newly acked or sacked when the sender receives an ACK. The current approach basically computes newly_acked_sacked = (prior_packets - prior_sacked) - (tp->packets_out - tp->sacked_out) where prior_packets and prior_sacked out are snapshot at the beginning of the ACK processing. The new approach tracks the delivery information via a new TCP state variable "delivered" which monotically increases as new packets are delivered in order or out-of-order. The reason for this change is that the current approach is brittle that produces negative or inaccurate estimate. 1) For non-SACK connections, an ACK that advances the SND.UNA could reset the DUPACK counters (tp->sacked_out) in tcp_process_loss() or tcp_fastretrans_alert(). This inflates the inflight suddenly and causes under-estimate or even negative estimate. Here is a real example: before after (processing ACK) packets_out 75 73 sacked_out 23 0 ca state Loss Open The old approach computes (75-23) - (73 - 0) = -21 delivered while the new approach computes 1 delivered since it considers the 2nd-24th packets are delivered OOO. 2) MSS change would re-count packets_out and sacked_out so the estimate is in-accurate and can even become negative. E.g., the inflight is doubled when MSS is halved. 3) Spurious retransmission signaled by DSACK is not accounted The new approach is simpler and more robust. For SACK connections, tp->delivered increments as packets are being acked or sacked in SACK and ACK processing. For non-sack connections, it's done in tcp_remove_reno_sacks() and tcp_add_reno_sack(). When an ACK advances the SND.UNA, tp->delivered is incremented by the number of packets ACKed (less the current number of DUPACKs received plus one packet hole). Upon receiving a DUPACK, tp->delivered is incremented assuming one out-of-order packet is delivered. Upon receiving a DSACK, tp->delivered is incremtened assuming one retransmission is delivered in tcp_sacktag_write_queue(). Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
Currently the cwnd is reduced and increased in various different places. The reduction happens in various places in the recovery state processing (tcp_fastretrans_alert) while the increase happens afterward. A better sequence is to identify lost packets and update the congestion control state (icsk_ca_state) first. Then base on the new state, up/down the cwnd in one central place. It's more clear to reason cwnd changes. Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Yuchung Cheng 提交于
The retransmission and F-RTO transmission currently happen inside recovery state processing (tcp_fastretrans_alert) but before congestion control. This refactoring moves the logic after both s.t. we can determine how much to send (cwnd) before deciding what to send. Signed-off-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <ncardwell@google.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 Yuchung Cheng 提交于
RFC 4015 section 3.4 says the TCP sender MUST refrain from reversing the congestion control state when the ACK signals congestion through the ECN-Echo flag. Currently we may not always do that when prior_ssthresh is reset upon receiving ACKs with ECE marks. This patch fixes 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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- 29 1月, 2016 1 次提交
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由 Neal Cardwell 提交于
This commit fixes a corner case in tcp_mark_head_lost() which was causing the WARN_ON(len > skb->len) in tcp_fragment() to fire. tcp_mark_head_lost() was assuming that if a packet has tcp_skb_pcount(skb) of N, then it's safe to fragment off a prefix of M*mss bytes, for any M < N. But with the tricky way TCP pcounts are maintained, this is not always true. For example, suppose the sender sends 4 1-byte packets and have the last 3 packet sacked. It will merge the last 3 packets in the write queue into an skb with pcount = 3 and len = 3 bytes. If another recovery happens after a sack reneging event, tcp_mark_head_lost() may attempt to split the skb assuming it has more than 2*MSS bytes. This sounds very counterintuitive, but as the commit description for the related commit c0638c24 ("tcp: don't fragment SACKed skbs in tcp_mark_head_lost()") notes, this is because tcp_shifted_skb() coalesces adjacent regions of SACKed skbs, and when doing this it preserves the sum of their packet counts in order to reflect the real-world dynamics on the wire. The c0638c24 commit tried to avoid problems by not fragmenting SACKed skbs, since SACKed skbs are where the non-proportionality between pcount and skb->len/mss is known to be possible. However, that commit did not handle the case where during a reneging event one of these weird SACKed skbs becomes an un-SACKed skb, which tcp_mark_head_lost() can then try to fragment. The fix is to simply mark the entire skb lost when this happens. This makes the recovery slightly more aggressive in such corner cases before we detect reordering. But once we detect reordering this code path is by-passed because FACK is disabled. 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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- 07 1月, 2016 1 次提交
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由 Yuchung Cheng 提交于
Patch 3759824d ("tcp: PRR uses CRB mode by default and SS mode conditionally") introduced a bug that cwnd may become 0 when both inflight and sndcnt are 0 (cwnd = inflight + sndcnt). This may lead to a div-by-zero if the connection starts another cwnd reduction phase by setting tp->prior_cwnd to the current cwnd (0) in tcp_init_cwnd_reduction(). To prevent this we skip PRR operation when nothing is acked or sacked. Then cwnd must be positive in all cases as long as ssthresh is positive: 1) The proportional reduction mode inflight > ssthresh > 0 2) The reduction bound mode a) inflight == ssthresh > 0 b) inflight < ssthresh sndcnt > 0 since newly_acked_sacked > 0 and inflight < ssthresh Therefore in all cases inflight and sndcnt can not both be 0. We check invalid tp->prior_cwnd to avoid potential div0 bugs. In reality this bug is triggered only with a sequence of less common events. For example, the connection is terminating an ECN-triggered cwnd reduction with an inflight 0, then it receives reordered/old ACKs or DSACKs from prior transmission (which acks nothing). Or the connection is in fast recovery stage that marks everything lost, but fails to retransmit due to local issues, then receives data packets from other end which acks nothing. Fixes: 3759824d ("tcp: PRR uses CRB mode by default and SS mode conditionally") Reported-by: NOleksandr Natalenko <oleksandr@natalenko.name> 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 12月, 2015 1 次提交
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由 David Ahern 提交于
Allow accepted sockets to derive their sk_bound_dev_if setting from the l3mdev domain in which the packets originated. A sysctl setting is added to control the behavior which is similar to sk_mark and sysctl_tcp_fwmark_accept. This effectively allow a process to have a "VRF-global" listen socket, with child sockets bound to the VRF device in which the packet originated. A similar behavior can be achieved using sk_mark, but a solution using marks is incomplete as it does not handle duplicate addresses in different L3 domains/VRFs. Allowing sockets to inherit the sk_bound_dev_if from l3mdev domain provides a complete solution. Signed-off-by: NDavid Ahern <dsa@cumulusnetworks.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 01 12月, 2015 1 次提交
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由 Eric Dumazet 提交于
Dmitry provided a syzkaller (http://github.com/google/syzkaller) generated program that triggers the WARNING at net/ipv4/tcp.c:1729 in tcp_recvmsg() : WARN_ON(tp->copied_seq != tp->rcv_nxt && !(flags & (MSG_PEEK | MSG_TRUNC))); His program is specifically attempting a Cross SYN TCP exchange, that we support (for the pleasure of hackers ?), but it looks we lack proper tcp->copied_seq initialization. Thanks again Dmitry for your report and testings. Signed-off-by: NEric Dumazet <edumazet@google.com> Reported-by: NDmitry Vyukov <dvyukov@google.com> Tested-by: NDmitry Vyukov <dvyukov@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 20 11月, 2015 1 次提交
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由 Eric Dumazet 提交于
tcp_send_rcvq() is used for re-injecting data into tcp receive queue. Problems : - No check against size is performed, allowed user to fool kernel in attempting very large memory allocations, eventually triggering OOM when memory is fragmented. - In case of fault during the copy we do not return correct errno. Lets use alloc_skb_with_frags() to cook optimal skbs. Fixes: 292e8d8c ("tcp: Move rcvq sending to tcp_input.c") Fixes: c0e88ff0 ("tcp: Repair socket queues") Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Pavel Emelyanov <xemul@parallels.com> Acked-by: NPavel Emelyanov <xemul@parallels.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 21 10月, 2015 6 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
a helper to prepare the main RACK patch 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
Remove the existing lost retransmit detection because RACK subsumes it completely. This also stops the overloading the ack_seq field of the skb control block. 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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由 Yuchung Cheng 提交于
Currently ca_seq_rtt_us does not use Kern's check. Fix that by checking if any packet acked is a retransmit, for both RTT used for RTT estimation and congestion control. Fixes: 5b08e47c ("tcp: prefer packet timing to TS-ECR for 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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- 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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