- 30 9月, 2014 3 次提交
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由 Florian Westphal 提交于
Suggested by Stephen. Also drop inline keyword and let compiler decide. gcc 4.7.3 decides to no longer inline tcp_ecn_check_ce, so split it up. The actual evaluation is not inlined anymore while the ECN_OK test is. Suggested-by: NStephen Hemminger <stephen@networkplumber.org> Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NFlorian Westphal <fw@strlen.de> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Florian Westphal 提交于
After Octavian Purdilas tcp ipv4/ipv6 unification work this helper only has a single callsite. While at it, convert name to lowercase, suggested by Stephen. Suggested-by: NStephen Hemminger <stephen@networkplumber.org> Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NFlorian Westphal <fw@strlen.de> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Li RongQing 提交于
This variable i is overwritten to 0 by following code Signed-off-by: NLi RongQing <roy.qing.li@gmail.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 29 9月, 2014 10 次提交
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由 Daniel Borkmann 提交于
This work adds the DataCenter TCP (DCTCP) congestion control algorithm [1], which has been first published at SIGCOMM 2010 [2], resp. follow-up analysis at SIGMETRICS 2011 [3] (and also, more recently as an informational IETF draft available at [4]). DCTCP is an enhancement to the TCP congestion control algorithm for data center networks. Typical data center workloads are i.e. i) partition/aggregate (queries; bursty, delay sensitive), ii) short messages e.g. 50KB-1MB (for coordination and control state; delay sensitive), and iii) large flows e.g. 1MB-100MB (data update; throughput sensitive). DCTCP has therefore been designed for such environments to provide/achieve the following three requirements: * High burst tolerance (incast due to partition/aggregate) * Low latency (short flows, queries) * High throughput (continuous data updates, large file transfers) with commodity, shallow buffered switches The basic idea of its design consists of two fundamentals: i) on the switch side, packets are being marked when its internal queue length > threshold K (K is chosen so that a large enough headroom for marked traffic is still available in the switch queue); ii) the sender/host side maintains a moving average of the fraction of marked packets, so each RTT, F is being updated as follows: F := X / Y, where X is # of marked ACKs, Y is total # of ACKs alpha := (1 - g) * alpha + g * F, where g is a smoothing constant The resulting alpha (iow: probability that switch queue is congested) is then being used in order to adaptively decrease the congestion window W: W := (1 - (alpha / 2)) * W The means for receiving marked packets resp. marking them on switch side in DCTCP is the use of ECN. RFC3168 describes a mechanism for using Explicit Congestion Notification from the switch for early detection of congestion, rather than waiting for segment loss to occur. However, this method only detects the presence of congestion, not the *extent*. In the presence of mild congestion, it reduces the TCP congestion window too aggressively and unnecessarily affects the throughput of long flows [4]. DCTCP, as mentioned, enhances Explicit Congestion Notification (ECN) processing to estimate the fraction of bytes that encounter congestion, rather than simply detecting that some congestion has occurred. DCTCP then scales the TCP congestion window based on this estimate [4], thus it can derive multibit feedback from the information present in the single-bit sequence of marks in its control law. And thus act in *proportion* to the extent of congestion, not its *presence*. Switches therefore set the Congestion Experienced (CE) codepoint in packets when internal queue lengths exceed threshold K. Resulting, DCTCP delivers the same or better throughput than normal TCP, while using 90% less buffer space. It was found in [2] that DCTCP enables the applications to handle 10x the current background traffic, without impacting foreground traffic. Moreover, a 10x increase in foreground traffic did not cause any timeouts, and thus largely eliminates TCP incast collapse problems. The algorithm itself has already seen deployments in large production data centers since then. We did a long-term stress-test and analysis in a data center, short summary of our TCP incast tests with iperf compared to cubic: This test measured DCTCP throughput and latency and compared it with CUBIC throughput and latency for an incast scenario. In this test, 19 senders sent at maximum rate to a single receiver. The receiver simply ran iperf -s. The senders ran iperf -c <receiver> -t 30. All senders started simultaneously (using local clocks synchronized by ntp). This test was repeated multiple times. Below shows the results from a single test. Other tests are similar. (DCTCP results were extremely consistent, CUBIC results show some variance induced by the TCP timeouts that CUBIC encountered.) For this test, we report statistics on the number of TCP timeouts, flow throughput, and traffic latency. 1) Timeouts (total over all flows, and per flow summaries): CUBIC DCTCP Total 3227 25 Mean 169.842 1.316 Median 183 1 Max 207 5 Min 123 0 Stddev 28.991 1.600 Timeout data is taken by measuring the net change in netstat -s "other TCP timeouts" reported. As a result, the timeout measurements above are not restricted to the test traffic, and we believe that it is likely that all of the "DCTCP timeouts" are actually timeouts for non-test traffic. We report them nevertheless. CUBIC will also include some non-test timeouts, but they are drawfed by bona fide test traffic timeouts for CUBIC. Clearly DCTCP does an excellent job of preventing TCP timeouts. DCTCP reduces timeouts by at least two orders of magnitude and may well have eliminated them in this scenario. 2) Throughput (per flow in Mbps): CUBIC DCTCP Mean 521.684 521.895 Median 464 523 Max 776 527 Min 403 519 Stddev 105.891 2.601 Fairness 0.962 0.999 Throughput data was simply the average throughput for each flow reported by iperf. By avoiding TCP timeouts, DCTCP is able to achieve much better per-flow results. In CUBIC, many flows experience TCP timeouts which makes flow throughput unpredictable and unfair. DCTCP, on the other hand, provides very clean predictable throughput without incurring TCP timeouts. Thus, the standard deviation of CUBIC throughput is dramatically higher than the standard deviation of DCTCP throughput. Mean throughput is nearly identical because even though cubic flows suffer TCP timeouts, other flows will step in and fill the unused bandwidth. Note that this test is something of a best case scenario for incast under CUBIC: it allows other flows to fill in for flows experiencing a timeout. Under situations where the receiver is issuing requests and then waiting for all flows to complete, flows cannot fill in for timed out flows and throughput will drop dramatically. 3) Latency (in ms): CUBIC DCTCP Mean 4.0088 0.04219 Median 4.055 0.0395 Max 4.2 0.085 Min 3.32 0.028 Stddev 0.1666 0.01064 Latency for each protocol was computed by running "ping -i 0.2 <receiver>" from a single sender to the receiver during the incast test. For DCTCP, "ping -Q 0x6 -i 0.2 <receiver>" was used to ensure that traffic traversed the DCTCP queue and was not dropped when the queue size was greater than the marking threshold. The summary statistics above are over all ping metrics measured between the single sender, receiver pair. The latency results for this test show a dramatic difference between CUBIC and DCTCP. CUBIC intentionally overflows the switch buffer which incurs the maximum queue latency (more buffer memory will lead to high latency.) DCTCP, on the other hand, deliberately attempts to keep queue occupancy low. The result is a two orders of magnitude reduction of latency with DCTCP - even with a switch with relatively little RAM. Switches with larger amounts of RAM will incur increasing amounts of latency for CUBIC, but not for DCTCP. 4) Convergence and stability test: This test measured the time that DCTCP took to fairly redistribute bandwidth when a new flow commences. It also measured DCTCP's ability to remain stable at a fair bandwidth distribution. DCTCP is compared with CUBIC for this test. At the commencement of this test, a single flow is sending at maximum rate (near 10 Gbps) to a single receiver. One second after that first flow commences, a new flow from a distinct server begins sending to the same receiver as the first flow. After the second flow has sent data for 10 seconds, the second flow is terminated. The first flow sends for an additional second. Ideally, the bandwidth would be evenly shared as soon as the second flow starts, and recover as soon as it stops. The results of this test are shown below. Note that the flow bandwidth for the two flows was measured near the same time, but not simultaneously. DCTCP performs nearly perfectly within the measurement limitations of this test: bandwidth is quickly distributed fairly between the two flows, remains stable throughout the duration of the test, and recovers quickly. CUBIC, in contrast, is slow to divide the bandwidth fairly, and has trouble remaining stable. CUBIC DCTCP Seconds Flow 1 Flow 2 Seconds Flow 1 Flow 2 0 9.93 0 0 9.92 0 0.5 9.87 0 0.5 9.86 0 1 8.73 2.25 1 6.46 4.88 1.5 7.29 2.8 1.5 4.9 4.99 2 6.96 3.1 2 4.92 4.94 2.5 6.67 3.34 2.5 4.93 5 3 6.39 3.57 3 4.92 4.99 3.5 6.24 3.75 3.5 4.94 4.74 4 6 3.94 4 5.34 4.71 4.5 5.88 4.09 4.5 4.99 4.97 5 5.27 4.98 5 4.83 5.01 5.5 4.93 5.04 5.5 4.89 4.99 6 4.9 4.99 6 4.92 5.04 6.5 4.93 5.1 6.5 4.91 4.97 7 4.28 5.8 7 4.97 4.97 7.5 4.62 4.91 7.5 4.99 4.82 8 5.05 4.45 8 5.16 4.76 8.5 5.93 4.09 8.5 4.94 4.98 9 5.73 4.2 9 4.92 5.02 9.5 5.62 4.32 9.5 4.87 5.03 10 6.12 3.2 10 4.91 5.01 10.5 6.91 3.11 10.5 4.87 5.04 11 8.48 0 11 8.49 4.94 11.5 9.87 0 11.5 9.9 0 SYN/ACK ECT test: This test demonstrates the importance of ECT on SYN and SYN-ACK packets by measuring the connection probability in the presence of competing flows for a DCTCP connection attempt *without* ECT in the SYN packet. The test was repeated five times for each number of competing flows. Competing Flows 1 | 2 | 4 | 8 | 16 ------------------------------ Mean Connection Probability 1 | 0.67 | 0.45 | 0.28 | 0 Median Connection Probability 1 | 0.65 | 0.45 | 0.25 | 0 As the number of competing flows moves beyond 1, the connection probability drops rapidly. Enabling DCTCP with this patch requires the following steps: DCTCP must be running both on the sender and receiver side in your data center, i.e.: sysctl -w net.ipv4.tcp_congestion_control=dctcp Also, ECN functionality must be enabled on all switches in your data center for DCTCP to work. The default ECN marking threshold (K) heuristic on the switch for DCTCP is e.g., 20 packets (30KB) at 1Gbps, and 65 packets (~100KB) at 10Gbps (K > 1/7 * C * RTT, [4]). In above tests, for each switch port, traffic was segregated into two queues. For any packet with a DSCP of 0x01 - or equivalently a TOS of 0x04 - the packet was placed into the DCTCP queue. All other packets were placed into the default drop-tail queue. For the DCTCP queue, RED/ECN marking was enabled, here, with a marking threshold of 75 KB. More details however, we refer you to the paper [2] under section 3). There are no code changes required to applications running in user space. DCTCP has been implemented in full *isolation* of the rest of the TCP code as its own congestion control module, so that it can run without a need to expose code to the core of the TCP stack, and thus nothing changes for non-DCTCP users. Changes in the CA framework code are minimal, and DCTCP algorithm operates on mechanisms that are already available in most Silicon. The gain (dctcp_shift_g) is currently a fixed constant (1/16) from the paper, but we leave the option that it can be chosen carefully to a different value by the user. In case DCTCP is being used and ECN support on peer site is off, DCTCP falls back after 3WHS to operate in normal TCP Reno mode. ss {-4,-6} -t -i diag interface: ... dctcp wscale:7,7 rto:203 rtt:2.349/0.026 mss:1448 cwnd:2054 ssthresh:1102 ce_state 0 alpha 15 ab_ecn 0 ab_tot 735584 send 10129.2Mbps pacing_rate 20254.1Mbps unacked:1822 retrans:0/15 reordering:101 rcv_space:29200 ... dctcp-reno wscale:7,7 rto:201 rtt:0.711/1.327 ato:40 mss:1448 cwnd:10 ssthresh:1102 fallback_mode send 162.9Mbps pacing_rate 325.5Mbps rcv_rtt:1.5 rcv_space:29200 More information about DCTCP can be found in [1-4]. [1] http://simula.stanford.edu/~alizade/Site/DCTCP.html [2] http://simula.stanford.edu/~alizade/Site/DCTCP_files/dctcp-final.pdf [3] http://simula.stanford.edu/~alizade/Site/DCTCP_files/dctcp_analysis-full.pdf [4] http://tools.ietf.org/html/draft-bensley-tcpm-dctcp-00 Joint work with Florian Westphal and Glenn Judd. Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NFlorian Westphal <fw@strlen.de> Signed-off-by: NGlenn Judd <glenn.judd@morganstanley.com> Acked-by: NStephen Hemminger <stephen@networkplumber.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Florian Westphal 提交于
DataCenter TCP (DCTCP) determines cwnd growth based on ECN information and ACK properties, e.g. ACK that updates window is treated differently than DUPACK. Also DCTCP needs information whether ACK was delayed ACK. Furthermore, DCTCP also implements a CE state machine that keeps track of CE markings of incoming packets. Therefore, extend the congestion control framework to provide these event types, so that DCTCP can be properly implemented as a normal congestion algorithm module outside of the core stack. Joint work with Daniel Borkmann and Glenn Judd. Signed-off-by: NFlorian Westphal <fw@strlen.de> Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NGlenn Judd <glenn.judd@morganstanley.com> Acked-by: NStephen Hemminger <stephen@networkplumber.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Florian Westphal 提交于
The congestion control ops "cwnd_event" currently supports CA_EVENT_FAST_ACK and CA_EVENT_SLOW_ACK events (among others). Both FAST and SLOW_ACK are only used by Westwood congestion control algorithm. This removes both flags from cwnd_event and adds a new in_ack_event callback for this. The goal is to be able to provide more detailed information about ACKs, such as whether ECE flag was set, or whether the ACK resulted in a window update. It is required for DataCenter TCP (DCTCP) congestion control algorithm as it makes a different choice depending on ECE being set or not. Joint work with Daniel Borkmann and Glenn Judd. Signed-off-by: NFlorian Westphal <fw@strlen.de> Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NGlenn Judd <glenn.judd@morganstanley.com> Acked-by: NStephen Hemminger <stephen@networkplumber.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Daniel Borkmann 提交于
This patch adds a flag to TCP congestion algorithms that allows for requesting to mark IPv4/IPv6 sockets with transport as ECN capable, that is, ECT(0), when required by a congestion algorithm. It is currently used and needed in DataCenter TCP (DCTCP), as it requires both peers to assert ECT on all IP packets sent - it uses ECN feedback (i.e. CE, Congestion Encountered information) from switches inside the data center to derive feedback to the end hosts. Therefore, simply add a new flag to icsk_ca_ops. Note that DCTCP's algorithm/behaviour slightly diverges from RFC3168, therefore this is only (!) enabled iff the assigned congestion control ops module has requested this. By that, we can tightly couple this logic really only to the provided congestion control ops. Joint work with Florian Westphal and Glenn Judd. Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NFlorian Westphal <fw@strlen.de> Signed-off-by: NGlenn Judd <glenn.judd@morganstanley.com> Acked-by: NStephen Hemminger <stephen@networkplumber.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Florian Westphal 提交于
Split assignment and initialization from one into two functions. This is required by followup patches that add Datacenter TCP (DCTCP) congestion control algorithm - we need to be able to determine if the connection is moderated by DCTCP before the 3WHS has finished. As we walk the available congestion control list during the assignment, we are always guaranteed to have Reno present as it's fixed compiled-in. Therefore, since we're doing the early assignment, we don't have a real use for the Reno alias tcp_init_congestion_ops anymore and can thus remove it. Actual usage of the congestion control operations are being made after the 3WHS has finished, in some cases however we can access get_info() via diag if implemented, therefore we need to zero out the private area for those modules. Joint work with Daniel Borkmann and Glenn Judd. Signed-off-by: NFlorian Westphal <fw@strlen.de> Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NGlenn Judd <glenn.judd@morganstanley.com> Acked-by: NStephen Hemminger <stephen@networkplumber.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Rick Jones 提交于
We do not wish to disturb dropwatch or perf drop profiles with an ARP we will ignore. Signed-off-by: NRick Jones <rick.jones2@hp.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Peter Pan(潘卫平) 提交于
This patch is a cleanup which follows the idea in commit e11ecddf (tcp: use TCP_SKB_CB(skb)->tcp_flags in input path), and it may reduce register pressure since skb->cb[] access is fast, bacause skb is probably in a register. v2: remove variable th v3: reword the changelog Signed-off-by: NWeiping Pan <panweiping3@gmail.com> Acked-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
Our goal is to access no more than one cache line access per skb in a write or receive queue when doing the various walks. After recent TCP_SKB_CB() reorganizations, it is almost done. Last part is tcp_skb_pcount() which currently uses skb_shinfo(skb)->gso_segs, which is a terrible choice, because it needs 3 cache lines in current kernel (skb->head, skb->end, and shinfo->gso_segs are all in 3 different cache lines, far from skb->cb) This very simple patch reuses space currently taken by tcp_tw_isn only in input path, as tcp_skb_pcount is only needed for skb stored in write queue. This considerably speeds up tcp_ack(), granted we avoid shinfo->tx_flags to get SKBTX_ACK_TSTAMP, which seems possible. This also speeds up all sack processing in general. This speeds up tcp_sendmsg() because it no longer has to access/dirty shinfo. 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
TCP maintains lists of skb in write queue, and in receive queues (in order and out of order queues) Scanning these lists both in input and output path usually requires access to skb->next, TCP_SKB_CB(skb)->seq, and TCP_SKB_CB(skb)->end_seq These fields are currently in two different cache lines, meaning we waste lot of memory bandwidth when these queues are big and flows have either packet drops or packet reorders. We can move TCP_SKB_CB(skb)->header at the end of TCP_SKB_CB, because this header is not used in fast path. This allows TCP to search much faster in the skb lists. Even with regular flows, we save one cache line miss in fast path. Thanks to Christoph Paasch for noticing we need to cleanup skb->cb[] (IPCB/IP6CB) before entering IP stack in tx path, and that I forgot IPCB use in tcp_v4_hnd_req() and tcp_v4_save_options(). 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
ip_options_echo() assumes struct ip_options is provided in &IPCB(skb)->opt Lets break this assumption, but provide a helper to not change all call points. ip_send_unicast_reply() gets a new struct ip_options pointer. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 27 9月, 2014 1 次提交
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由 Eric Dumazet 提交于
While profiling TCP stack, I noticed one useless atomic operation in tcp_sendmsg(), caused by skb_header_release(). It turns out all current skb_header_release() users have a fresh skb, that no other user can see, so we can avoid one atomic operation. Introduce __skb_header_release() to clearly document this. This gave me a 1.5 % improvement on TCP_RR workload. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 26 9月, 2014 3 次提交
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由 Tom Herbert 提交于
The send_check logic was only interesting in cases of TCP offload and UDP UFO where the checksum needed to be initialized to the pseudo header checksum. Now we've moved that logic into the related gso_segment functions so gso_send_check is no longer needed. Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Tom Herbert 提交于
In udp[46]_ufo_send_check the UDP checksum initialized to the pseudo header checksum. We can move this logic into udp[46]_ufo_fragment. After this change udp[64]_ufo_send_check is a no-op. Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Tom Herbert 提交于
In tcp_v[46]_gso_send_check the TCP checksum is initialized to the pseudo header checksum using __tcp_v[46]_send_check. We can move this logic into new tcp[46]_gso_segment functions to be done when ip_summed != CHECKSUM_PARTIAL (ip_summed == CHECKSUM_PARTIAL should be the common case, possibly always true when taking GSO path). After this change tcp_v[46]_gso_send_check is no-op. Signed-off-by: NTom Herbert <therbert@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 24 9月, 2014 2 次提交
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由 Eric Dumazet 提交于
In order to make TCP more resilient in presence of reorders, we need to allow coalescing to happen when skbs from out of order queue are transferred into receive queue. LRO/GRO can be completely canceled in some pathological cases, like per packet load balancing on aggregated links. I had to move tcp_try_coalesce() up in the file above tcp_ofo_queue() 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
Current ICMP rate limiting uses inetpeer cache, which is an RBL tree protected by a lock, meaning that hosts can be stuck hard if all cpus want to check ICMP limits. When say a DNS or NTP server process is restarted, inetpeer tree grows quick and machine comes to its knees. iptables can not help because the bottleneck happens before ICMP messages are even cooked and sent. This patch adds a new global limitation, using a token bucket filter, controlled by two new sysctl : icmp_msgs_per_sec - INTEGER Limit maximal number of ICMP packets sent per second from this host. Only messages whose type matches icmp_ratemask are controlled by this limit. Default: 1000 icmp_msgs_burst - INTEGER icmp_msgs_per_sec controls number of ICMP packets sent per second, while icmp_msgs_burst controls the burst size of these packets. Default: 50 Note that if we really want to send millions of ICMP messages per second, we might extend idea and infra added in commit 04ca6973 ("ip: make IP identifiers less predictable") : add a token bucket in the ip_idents hash and no longer rely on inetpeer. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 23 9月, 2014 2 次提交
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由 Eric Dumazet 提交于
this_cpu_ptr() in preemptible context is generally bad Sep 22 05:05:55 br kernel: [ 94.608310] BUG: using smp_processor_id() in preemptible [00000000] code: ip/2261 Sep 22 05:05:55 br kernel: [ 94.608316] caller is tunnel_dst_set.isra.28+0x20/0x60 [ip_tunnel] Sep 22 05:05:55 br kernel: [ 94.608319] CPU: 3 PID: 2261 Comm: ip Not tainted 3.17.0-rc5 #82 We can simply use raw_cpu_ptr(), as preemption is safe in these contexts. Should fix https://bugzilla.kernel.org/show_bug.cgi?id=84991Signed-off-by: NEric Dumazet <edumazet@google.com> Reported-by: NJoe <joe9mail@gmail.com> Fixes: 9a4aa9af ("ipv4: Use percpu Cache route in IP tunnels") Acked-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
icsk_rto is a 32bit field, and icsk_backoff can reach 15 by default, or more if some sysctl (eg tcp_retries2) are changed. Better use 64bit to perform icsk_rto << icsk_backoff operations As Joe Perches suggested, add a helper for this. Yuchung spotted the tcp_v4_err() case. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 20 9月, 2014 9 次提交
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由 Tom Herbert 提交于
Added netlink attrs to configure FOU encapsulation for GRE, netlink handling of these flags, and properly adjust MTU for encapsulation. ip_tunnel_encap is called from ip_tunnel_xmit to actually perform FOU encapsulation. Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Tom Herbert 提交于
Add netlink handling for IP tunnel encapsulation parameters and and adjustment of MTU for encapsulation. ip_tunnel_encap is called from ip_tunnel_xmit to actually perform FOU encapsulation. Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Tom Herbert 提交于
This patch changes IP tunnel to support (secondary) encapsulation, Foo-over-UDP. Changes include: 1) Adding tun_hlen as the tunnel header length, encap_hlen as the encapsulation header length, and hlen becomes the grand total of these. 2) Added common netlink define to support FOU encapsulation. 3) Routines to perform FOU encapsulation. Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Tom Herbert 提交于
Implement fou_gro_receive and fou_gro_complete, and populate these in the correponsing udp_offloads for the socket. Added ipproto to udp_offloads and pass this from UDP to the fou GRO routine in proto field of napi_gro_cb structure. Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Tom Herbert 提交于
This patch provides a receive path for foo-over-udp. This allows direct encapsulation of IP protocols over UDP. The bound destination port is used to map to an IP protocol, and the XFRM framework (udp_encap_rcv) is used to receive encapsulated packets. Upon reception, the encapsulation header is logically removed (pointer to transport header is advanced) and the packet is reinjected into the receive path with the IP protocol indicated by the mapping. Netlink is used to configure FOU ports. The configuration information includes the port number to bind to and the IP protocol corresponding to that port. This should support GRE/UDP (http://tools.ietf.org/html/draft-yong-tsvwg-gre-in-udp-encap-02), as will as the other IP tunneling protocols (IPIP, SIT). Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Tom Herbert 提交于
Want to be able to use these in foo-over-udp offloads, etc. Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
Now we no longer rely on having tcp headers for skbs in receive queue, tcp repair do not need to build fake ones. Signed-off-by: NEric Dumazet <edumazet@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Andy Zhou 提交于
Added a few more UDP tunnel APIs that can be shared by UDP based tunnel protocol implementation. The main ones are highlighted below. setup_udp_tunnel_sock() configures UDP listener socket for receiving UDP encapsulated packets. udp_tunnel_xmit_skb() and upd_tunnel6_xmit_skb() transmit skb using UDP encapsulation. udp_tunnel_sock_release() closes the UDP tunnel listener socket. Signed-off-by: NAndy Zhou <azhou@nicira.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Andy Zhou 提交于
Add ip6_udp_tunnel.c for ipv6 UDP tunnel functions to avoid ifdefs in udp_tunnel.c Signed-off-by: NAndy Zhou <azhou@nicira.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 18 9月, 2014 1 次提交
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由 Herbert Xu 提交于
While tracking down the MAX_AH_AUTH_LEN crash in an old kernel I thought that this limit was rather arbitrary and we should just get rid of it. In fact it seems that we've already done all the work needed to remove it apart from actually removing it. This limit was there in order to limit stack usage. Since we've already switched over to allocating scratch space using kmalloc, there is no longer any need to limit the authentication length. This patch kills all references to it, including the BUG_ONs that led me here. Signed-off-by: NHerbert Xu <herbert@gondor.apana.org.au> Signed-off-by: NSteffen Klassert <steffen.klassert@secunet.com>
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- 16 9月, 2014 4 次提交
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由 Steffen Klassert 提交于
Currently we genarate a blackhole route route whenever we have matching policies but can not resolve the states. Here we assume that dst_output() is called to kill the balckholed packets. Unfortunately this assumption is not true in all cases, so it is possible that these packets leave the system unwanted. We fix this by generating blackhole routes only from the route lookup functions, here we can guarantee a call to dst_output() afterwards. Fixes: 2774c131 ("xfrm: Handle blackhole route creation via afinfo.") Reported-by: NKonstantinos Kolelis <k.kolelis@sirrix.com> Signed-off-by: NSteffen Klassert <steffen.klassert@secunet.com>
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由 Eric Dumazet 提交于
tcp_collapse() wants to shrink skb so that the overhead is minimal. Now we store tcp flags into TCP_SKB_CB(skb)->tcp_flags, we no longer need to keep around full headers. Whole available space is dedicated to the payload. Signed-off-by: NEric Dumazet <edumazet@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
We can allow a segment with FIN to be aggregated, if we take care to add tcp flags, and if skb_try_coalesce() takes care of zero sized skbs. Signed-off-by: NEric Dumazet <edumazet@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
Input path of TCP do not currently uses TCP_SKB_CB(skb)->tcp_flags, which is only used in output path. tcp_recvmsg(), looks at tcp_hdr(skb)->syn for every skb found in receive queue, and its unfortunate because this bit is located in a cache line right before the payload. We can simplify TCP by copying tcp flags into TCP_SKB_CB(skb)->tcp_flags. This patch does so, and avoids the cache line miss in tcp_recvmsg() Following patches will - allow a segment with FIN being coalesced in tcp_try_coalesce() - simplify tcp_collapse() by not copying the headers. Signed-off-by: NEric Dumazet <edumazet@google.com> Acked-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 13 9月, 2014 1 次提交
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由 Scott Wood 提交于
Commit 2abb7cdc ("udp: Add support for doing checksum unnecessary conversion") caused napi_gro_cb structs with the "flush" field zero to take the "udp_gro_receive" path rather than the "set flush to 1" path that they would previously take. As a result I saw booting from an NFS root hang shortly after starting userspace, with "server not responding" messages. This change to the handling of "flush == 0" packets appears to be incidental to the goal of adding new code in the case where skb_gro_checksum_validate_zero_check() returns zero. Based on that and the fact that it breaks things, I'm assuming that it is unintentional. Fixes: 2abb7cdc ("udp: Add support for doing checksum unnecessary conversion") Cc: Tom Herbert <therbert@google.com> Signed-off-by: NScott Wood <scottwood@freescale.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 10 9月, 2014 4 次提交
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由 Tom Herbert 提交于
Add inet_gro_receive and inet_gro_complete to ipip_offload to support GRO. Signed-off-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
net/ipv4/udp_offload.c:339:5: warning: symbol 'udp4_gro_complete' was not declared. Should it be static? Signed-off-by: NEric Dumazet <edumazet@google.com> Cc: Tom Herbert <therbert@google.com> Fixes: 57c67ff4 ("udp: additional GRO support") Acked-by: NTom Herbert <therbert@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Eric Dumazet 提交于
Remove one sparse warning : net/ipv4/ip_sockglue.c:328:22: warning: incorrect type in assignment (different address spaces) net/ipv4/ip_sockglue.c:328:22: expected struct ip_ra_chain [noderef] <asn:4>*next net/ipv4/ip_sockglue.c:328:22: got struct ip_ra_chain *[assigned] ra And replace one rcu_assign_ptr() by RCU_INIT_POINTER() where applicable. 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
Alexander Duyck reported high false sharing on dst refcount in tcp stack when prequeue is used. prequeue is the mechanism used when a thread is blocked in recvmsg()/read() on a TCP socket, using a blocking model rather than select()/poll()/epoll() non blocking one. We already try to use RCU in input path as much as possible, but we were forced to take a refcount on the dst when skb escaped RCU protected region. When/if the user thread runs on different cpu, dst_release() will then touch dst refcount again. Commit 09316255 (tcp: force a dst refcount when prequeue packet) was an example of a race fix. It turns out the only remaining usage of skb->dst for a packet stored in a TCP socket prequeue is IP early demux. We can add a logic to detect when IP early demux is probably going to use skb->dst. Because we do an optimistic check rather than duplicate existing logic, we need to guard inet_sk_rx_dst_set() and inet6_sk_rx_dst_set() from using a NULL dst. Many thanks to Alexander for providing a nice bug report, git bisection, and reproducer. Tested using Alexander script on a 40Gb NIC, 8 RX queues. Hosts have 24 cores, 48 hyper threads. echo 0 >/proc/sys/net/ipv4/tcp_autocorking for i in `seq 0 47` do for j in `seq 0 2` do netperf -H $DEST -t TCP_STREAM -l 1000 \ -c -C -T $i,$i -P 0 -- \ -m 64 -s 64K -D & done done Before patch : ~6Mpps and ~95% cpu usage on receiver After patch : ~9Mpps and ~35% cpu usage on receiver. Signed-off-by: NEric Dumazet <edumazet@google.com> Reported-by: NAlexander Duyck <alexander.h.duyck@intel.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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