- 25 3月, 2015 3 次提交
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由 Eric Dumazet 提交于
With request socks convergence, we no longer need different lookup methods. A request socket can use generic lookup function. Add const qualifier to 2nd tcp_v[46]_md5_lookup() parameter. 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
Since request and established sockets now have same base, there is no need to pass two pointers to tcp_v4_md5_hash_skb() or tcp_v6_md5_hash_skb() Also add a const qualifier to their struct tcp_md5sig_key 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
While timer handler effectively runs a rcu read locked section, there is no explicit rcu_read_lock()/rcu_read_unlock() annotations and lockdep can be confused here : net/ipv4/tcp_ipv4.c-906- /* caller either holds rcu_read_lock() or socket lock */ net/ipv4/tcp_ipv4.c:907: md5sig = rcu_dereference_check(tp->md5sig_info, net/ipv4/tcp_ipv4.c-908- sock_owned_by_user(sk) || net/ipv4/tcp_ipv4.c-909- lockdep_is_held(&sk->sk_lock.slock)); Let's explicitely acquire rcu_read_lock() in tcp_make_synack() Before commit fa76ce73 ("inet: get rid of central tcp/dccp listener timer"), we were holding listener lock so lockdep was happy. Fixes: fa76ce73 ("inet: get rid of central tcp/dccp listener timer") Signed-off-by: NEric DUmazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 21 3月, 2015 2 次提交
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由 Eric Dumazet 提交于
This reverts commit ca10b9e9. No longer needed after commit eb8895de ("tcp: tcp_make_synack() should use sock_wmalloc") When under SYNFLOOD, we build lot of SYNACK and hit false sharing because of multiple modifications done on sk_listener->sk_wmem_alloc Since tcp_make_synack() uses sock_wmalloc(), there is no need to call skb_set_owner_w() again, as this adds two atomic operations. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Josh Hunt 提交于
tcp_send_fin() does not account for the memory it allocates properly, so sk_forward_alloc can be negative in cases where we've sent a FIN: ss example output (ss -amn | grep -B1 f4294): tcp FIN-WAIT-1 0 1 192.168.0.1:45520 192.0.2.1:8080 skmem:(r0,rb87380,t0,tb87380,f4294966016,w1280,o0,bl0) Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 07 3月, 2015 2 次提交
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由 Fan Du 提交于
As per RFC4821 7.3. Selecting Probe Size, a probe timer should be armed once probing has converged. Once this timer expired, probing again to take advantage of any path PMTU change. The recommended probing interval is 10 minutes per RFC1981. Probing interval could be sysctled by sysctl_tcp_probe_interval. Eric Dumazet suggested to implement pseudo timer based on 32bits jiffies tcp_time_stamp instead of using classic timer for such rare event. Signed-off-by: NFan Du <fan.du@intel.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Fan Du 提交于
Current probe_size is chosen by doubling mss_cache, the probing process will end shortly with a sub-optimal mss size, and the link mtu will not be taken full advantage of, in return, this will make user to tweak tcp_base_mss with care. Use binary search to choose probe_size in a fine granularity manner, an optimal mss will be found to boost performance as its maxmium. In addition, introduce a sysctl_tcp_probe_threshold to control when probing will stop in respect to the width of search range. Test env: Docker instance with vxlan encapuslation(82599EB) iperf -c 10.0.0.24 -t 60 before this patch: 1.26 Gbits/sec After this patch: increase 26% 1.59 Gbits/sec Signed-off-by: NFan Du <fan.du@intel.com> Acked-by: NJohn Heffner <johnwheffner@gmail.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 01 3月, 2015 3 次提交
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由 Eric Dumazet 提交于
Another TCP issue is triggered by ECN. Under pressure, receiver gets ECN marks, and send back ACK packets with ECE TCP flag. Senders enter CA_CWR state. In this state, tcp_tso_should_defer() is short cut : if (icsk->icsk_ca_state != TCP_CA_Open) goto send_now; This means that about all ACK packets we receive are triggering a partial send, and because cwnd is kept small, we can only send a small amount of data for each incoming ACK, which in return generate more ACK packets. Allowing CA_Open and CA_CWR states to enable TSO defer in tcp_tso_should_defer() brings performance back : TSO autodefer has more chance to defer under pressure. This patch increases TSO and LRO/GRO efficiency back to normal levels, and does not impact overall ECN behavior. 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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由 Eric Dumazet 提交于
With sysctl_tcp_min_tso_segs being 4, it is very possible that tcp_tso_should_defer() decides not sending last 2 MSS of initial window of 10 packets. This also applies if autosizing decides to send X MSS per GSO packet, and cwnd is not a multiple of X. This patch implements an heuristic based on age of first skb in write queue : If it was sent very recently (less than half srtt), we can predict that no ACK packet will come in less than half rtt, so deferring might cause an under utilization of our window. This is visible on initial send (IW10) on web servers, but more generally on some RPC, as the last part of the message might need an extra RTT to get delivered. Tested: Ran following packetdrill test // A simple server-side test that sends exactly an initial window (IW10) // worth of packets. `sysctl -e -q net.ipv4.tcp_min_tso_segs=4` 0.000 socket(..., SOCK_STREAM, IPPROTO_TCP) = 3 +0 setsockopt(3, SOL_SOCKET, SO_REUSEADDR, [1], 4) = 0 +0 bind(3, ..., ...) = 0 +0 listen(3, 1) = 0 +.1 < S 0:0(0) win 32792 <mss 1460,sackOK,nop,nop,nop,wscale 7> +0 > S. 0:0(0) ack 1 <mss 1460,nop,nop,sackOK,nop,wscale 6> +.1 < . 1:1(0) ack 1 win 257 +0 accept(3, ..., ...) = 4 +0 write(4, ..., 14600) = 14600 +0 > . 1:5841(5840) ack 1 win 457 +0 > . 5841:11681(5840) ack 1 win 457 // Following packet should be sent right now. +0 > P. 11681:14601(2920) ack 1 win 457 +.1 < . 1:1(0) ack 14601 win 257 +0 close(4) = 0 +0 > F. 14601:14601(0) ack 1 +.1 < F. 1:1(0) ack 14602 win 257 +0 > . 14602:14602(0) ack 2 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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由 Eric Dumazet 提交于
TSO relies on ability to defer sending a small amount of packets. Heuristic is to wait for future ACKS in hope to send more packets at once. Current algorithm uses a per socket tso_deferred field as a pseudo timer. This pseudo timer relies on future ACK, but there is no guarantee we receive them in time. Fix would be to use a real timer, but cost of such timer is probably too expensive for typical cases. This patch changes the logic to test the time of last transmit, because we should not add bursts of more than 1ms for any given flow. We've used this patch for about two years at Google, before FQ/pacing as it would reduce a fair amount of bursts. 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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- 10 2月, 2015 1 次提交
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由 Fan Du 提交于
Packetization Layer Path MTU Discovery works separately beside Path MTU Discovery at IP level, different net namespace has various requirements on which one to chose, e.g., a virutalized container instance would require TCP PMTU to probe an usable effective mtu for underlying tunnel, while the host would employ classical ICMP based PMTU to function. Hence making TCP PMTU mechanism per net namespace to decouple two functionality. Furthermore the probe base MSS should also be configured separately for each namespace. Signed-off-by: NFan Du <fan.du@intel.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 05 2月, 2015 1 次提交
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由 Eric Dumazet 提交于
When we added pacing to TCP, we decided to let sch_fq take care of actual pacing. All TCP had to do was to compute sk->pacing_rate using simple formula: sk->pacing_rate = 2 * cwnd * mss / rtt It works well for senders (bulk flows), but not very well for receivers or even RPC : cwnd on the receiver can be less than 10, rtt can be around 100ms, so we can end up pacing ACK packets, slowing down the sender. Really, only the sender should pace, according to its own logic. Instead of adding a new bit in skb, or call yet another flow dissection, we tweak skb->truesize to a small value (2), and we instruct sch_fq to use new helper and not pace pure ack. Note this also helps TCP small queue, as ack packets present in qdisc/NIC do not prevent sending a data packet (RPC workload) This helps to reduce tx completion overhead, ack packets can use regular sock_wfree() instead of tcp_wfree() which is a bit more expensive. This has no impact in the case packets are sent to loopback interface, as we do not coalesce ack packets (were we would detect skb->truesize lie) In case netem (with a delay) is used, skb_orphan_partial() also sets skb->truesize to 1. This patch is a combination of two patches we used for about one year at Google. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 04 2月, 2015 1 次提交
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由 Al Viro 提交于
patch is actually smaller than it seems to be - most of it is unindenting the inner loop body in tcp_sendmsg() itself... the bit in tcp_input.c is going to get reverted very soon - that's what memcpy_from_msg() will become, but not in this commit; let's keep it reasonably contained... There's one potentially subtle change here: in case of short copy from userland, mainline tcp_send_syn_data() discards the skb it has allocated and falls back to normal path, where we'll send as much as possible after rereading the same data again. This patch trims SYN+data skb instead - that way we don't need to copy from the same place twice. Signed-off-by: NAl Viro <viro@zeniv.linux.org.uk>
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- 06 1月, 2015 1 次提交
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由 Daniel Borkmann 提交于
This work adds the possibility to define a per route/destination congestion control algorithm. Generally, this opens up the possibility for a machine with different links to enforce specific congestion control algorithms with optimal strategies for each of them based on their network characteristics, even transparently for a single application listening on all links. For our specific use case, this additionally facilitates deployment of DCTCP, for example, applications can easily serve internal traffic/dsts in DCTCP and external one with CUBIC. Other scenarios would also allow for utilizing e.g. long living, low priority background flows for certain destinations/routes while still being able for normal traffic to utilize the default congestion control algorithm. We also thought about a per netns setting (where different defaults are possible), but given its actually a link specific property, we argue that a per route/destination setting is the most natural and flexible. The administrator can utilize this through ip-route(8) by appending "congctl [lock] <name>", where <name> denotes the name of a congestion control algorithm and the optional lock parameter allows to enforce the given algorithm so that applications in user space would not be allowed to overwrite that algorithm for that destination. The dst metric lookups are being done when a dst entry is already available in order to avoid a costly lookup and still before the algorithms are being initialized, thus overhead is very low when the feature is not being used. While the client side would need to drop the current reference on the module, on server side this can actually even be avoided as we just got a flat-copied socket clone. Joint work with Florian Westphal. Suggested-by: NHannes Frederic Sowa <hannes@stressinduktion.org> Signed-off-by: NFlorian Westphal <fw@strlen.de> Signed-off-by: NDaniel Borkmann <dborkman@redhat.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 03 1月, 2015 1 次提交
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由 Herbert Xu 提交于
Thomas Jarosch reported IPsec TCP stalls when a PMTU event occurs. In fact the problem was completely unrelated to IPsec. The bug is also reproducible if you just disable TSO/GSO. The problem is that when the MSS goes down, existing queued packet on the TX queue that have not been transmitted yet all look like TSO packets and get treated as such. This then triggers a bug where tcp_mss_split_point tells us to generate a zero-sized packet on the TX queue. Once that happens we're screwed because the zero-sized packet can never be removed by ACKs. Fixes: 1485348d ("tcp: Apply device TSO segment limit earlier") Reported-by: NThomas Jarosch <thomas.jarosch@intra2net.com> Signed-off-by: NHerbert Xu <herbert@gondor.apana.org.au> Cheers, Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 10 12月, 2014 2 次提交
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由 Eric Dumazet 提交于
Commit 95bd09eb ("tcp: TSO packets automatic sizing") tried to control TSO size, but did this at the wrong place (sendmsg() time) At sendmsg() time, we might have a pessimistic view of flow rate, and we end up building very small skbs (with 2 MSS per skb). This is bad because : - It sends small TSO packets even in Slow Start where rate quickly increases. - It tends to make socket write queue very big, increasing tcp_ack() processing time, but also increasing memory needs, not necessarily accounted for, as fast clones overhead is currently ignored. - Lower GRO efficiency and more ACK packets. Servers with a lot of small lived connections suffer from this. Lets instead fill skbs as much as possible (64KB of payload), but split them at xmit time, when we have a precise idea of the flow rate. skb split is actually quite efficient. Patch looks bigger than necessary, because TCP Small Queue decision now has to take place after the eventual split. As Neal suggested, introduce a new tcp_tso_autosize() helper, so that tcp_tso_should_defer() can be synchronized on same goal. Rename tp->xmit_size_goal_segs to tp->gso_segs, as this variable contains number of mss that we can put in GSO packet, and is not related to the autosizing goal anymore. Tested: 40 ms rtt link nstat >/dev/null netperf -H remote -l -2000000 -- -s 1000000 nstat | egrep "IpInReceives|IpOutRequests|TcpOutSegs|IpExtOutOctets" Before patch : Recv Send Send Socket Socket Message Elapsed Size Size Size Time Throughput bytes bytes bytes secs. 10^6bits/s 87380 2000000 2000000 0.36 44.22 IpInReceives 600 0.0 IpOutRequests 599 0.0 TcpOutSegs 1397 0.0 IpExtOutOctets 2033249 0.0 After patch : Recv Send Send Socket Socket Message Elapsed Size Size Size Time Throughput bytes bytes bytes secs. 10^6bits/sec 87380 2000000 2000000 0.36 44.27 IpInReceives 221 0.0 IpOutRequests 232 0.0 TcpOutSegs 1397 0.0 IpExtOutOctets 2013953 0.0 Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NNeal Cardwell <ncardwell@google.com> Acked-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Al Viro 提交于
Note that the code _using_ ->msg_iter at that point will be very unhappy with anything other than unshifted iovec-backed iov_iter. We still need to convert users to proper primitives. Signed-off-by: NAl Viro <viro@zeniv.linux.org.uk>
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- 20 11月, 2014 1 次提交
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由 Eric Dumazet 提交于
While working on sk_forward_alloc problems reported by Denys Fedoryshchenko, we found that tcp connect() (and fastopen) do not call sk_wmem_schedule() for SYN packet (and/or SYN/DATA packet), so sk_forward_alloc is negative while connect is in progress. We can fix this by calling regular sk_stream_alloc_skb() both for the SYN packet (in tcp_connect()) and the syn_data packet in tcp_send_syn_data() Then, tcp_send_syn_data() can avoid copying syn_data as we simply can manipulate syn_data->cb[] to remove SYN flag (and increment seq) Instead of open coding memcpy_fromiovecend(), simply use this helper. This leaves in socket write queue clean fast clone skbs. This was tested against our fastopen packetdrill tests. Reported-by: NDenys Fedoryshchenko <nuclearcat@nuclearcat.com> 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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- 14 11月, 2014 1 次提交
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由 Eric Dumazet 提交于
In DC world, GSO packets initially cooked by tcp_sendmsg() are usually big, as sk_pacing_rate is high. When network is congested, cwnd can be smaller than the GSO packets found in socket write queue. tcp_write_xmit() splits GSO packets using the available cwnd, and we end up sending a single GSO packet, consuming all available cwnd. With GRO aggregation on the receiver, we might handle a single GRO packet, sending back a single ACK. 1) This single ACK might be lost TLP or RTO are forced to attempt a retransmit. 2) This ACK releases a full cwnd, sender sends another big GSO packet, in a ping pong mode. This behavior does not fill the pipes in the best way, because of scheduling artifacts. Make sure we always have at least two GSO packets in flight. This allows us to safely increase GRO efficiency without risking spurious retransmits. 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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- 05 11月, 2014 1 次提交
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由 Florian Westphal 提交于
This patch allows to set ECN on a per-route basis in case the sysctl tcp_ecn is not set to 1. In other words, when ECN is set for specific routes, it provides a tcp_ecn=1 behaviour for that route while the rest of the stack acts according to the global settings. One can use 'ip route change dev $dev $net features ecn' to toggle this. Having a more fine-grained per-route setting can be beneficial for various reasons, for example, 1) within data centers, or 2) local ISPs may deploy ECN support for their own video/streaming services [1], etc. There was a recent measurement study/paper [2] which scanned the Alexa's publicly available top million websites list from a vantage point in US, Europe and Asia: Half of the Alexa list will now happily use ECN (tcp_ecn=2, most likely blamed to commit 255cac91 ("tcp: extend ECN sysctl to allow server-side only ECN") ;)); the break in connectivity on-path was found is about 1 in 10,000 cases. Timeouts rather than receiving back RSTs were much more common in the negotiation phase (and mostly seen in the Alexa middle band, ranks around 50k-150k): from 12-thousand hosts on which there _may_ be ECN-linked connection failures, only 79 failed with RST when _not_ failing with RST when ECN is not requested. It's unclear though, how much equipment in the wild actually marks CE when buffers start to fill up. We thought about a fallback to non-ECN for retransmitted SYNs as another global option (which could perhaps one day be made default), but as Eric points out, there's much more work needed to detect broken middleboxes. Two examples Eric mentioned are buggy firewalls that accept only a single SYN per flow, and middleboxes that successfully let an ECN flow establish, but later mark CE for all packets (so cwnd converges to 1). [1] http://www.ietf.org/proceedings/89/slides/slides-89-tsvarea-1.pdf, p.15 [2] http://ecn.ethz.ch/ Joint work with Daniel Borkmann. Reference: http://thread.gmane.org/gmane.linux.network/335797Suggested-by: NHannes Frederic Sowa <hannes@stressinduktion.org> Acked-by: NEric Dumazet <edumazet@google.com> 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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- 31 10月, 2014 1 次提交
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由 Eric Dumazet 提交于
Some drivers are unable to perform TX completions in a bound time. They instead call skb_orphan() Problem is skb_fclone_busy() has to detect this case, otherwise we block TCP retransmits and can freeze unlucky tcp sessions on mostly idle hosts. Signed-off-by: NEric Dumazet <edumazet@google.com> Fixes: 1f3279ae ("tcp: avoid retransmits of TCP packets hanging in host queues") Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 15 10月, 2014 2 次提交
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由 Eric Dumazet 提交于
TCP Small queues tries to keep number of packets in qdisc as small as possible, and depends on a tasklet to feed following packets at TX completion time. Choice of tasklet was driven by latencies requirements. Then, TCP stack tries to avoid reorders, by locking flows with outstanding packets in qdisc in a given TX queue. What can happen is that many flows get attracted by a low performing TX queue, and cpu servicing TX completion has to feed packets for all of them, making this cpu 100% busy in softirq mode. This became particularly visible with latest skb->xmit_more support Strategy adopted in this patch is to detect when tcp_wfree() is called from ksoftirqd and let the outstanding queue for this flow being drained before feeding additional packets, so that skb->ooo_okay can be set to allow select_queue() to select the optimal queue : Incoming ACKS are normally handled by different cpus, so this patch gives more chance for these cpus to take over the burden of feeding qdisc with future packets. Tested: lpaa23:~# ./super_netperf 1400 --google-pacing-rate 3028000 -H lpaa24 -l 3600 & lpaa23:~# sar -n DEV 1 10 | grep eth1 06:16:18 AM eth1 595448.00 1190564.00 38381.09 1760253.12 0.00 0.00 1.00 06:16:19 AM eth1 594858.00 1189686.00 38340.76 1758952.72 0.00 0.00 0.00 06:16:20 AM eth1 597017.00 1194019.00 38480.79 1765370.29 0.00 0.00 1.00 06:16:21 AM eth1 595450.00 1190936.00 38380.19 1760805.05 0.00 0.00 0.00 06:16:22 AM eth1 596385.00 1193096.00 38442.56 1763976.29 0.00 0.00 1.00 06:16:23 AM eth1 598155.00 1195978.00 38552.97 1768264.60 0.00 0.00 0.00 06:16:24 AM eth1 594405.00 1188643.00 38312.57 1757414.89 0.00 0.00 1.00 06:16:25 AM eth1 593366.00 1187154.00 38252.16 1755195.83 0.00 0.00 0.00 06:16:26 AM eth1 593188.00 1186118.00 38232.88 1753682.57 0.00 0.00 1.00 06:16:27 AM eth1 596301.00 1192241.00 38440.94 1762733.09 0.00 0.00 0.00 Average: eth1 595457.30 1190843.50 38381.69 1760664.84 0.00 0.00 0.50 lpaa23:~# ./tc -s -d qd sh dev eth1 | grep backlog backlog 7606336b 2513p requeues 167982 backlog 224072b 74p requeues 566 backlog 581376b 192p requeues 5598 backlog 181680b 60p requeues 1070 backlog 5305056b 1753p requeues 110166 // Here, this TX queue is attracting flows backlog 157456b 52p requeues 1758 backlog 672216b 222p requeues 3025 backlog 60560b 20p requeues 24541 backlog 448144b 148p requeues 21258 lpaa23:~# echo 1 >/proc/sys/net/ipv4/tcp_tsq_enable_tcp_wfree_ksoftirqd_detect Immediate jump to full bandwidth, and traffic is properly shard on all tx queues. lpaa23:~# sar -n DEV 1 10 | grep eth1 06:16:46 AM eth1 1397632.00 2795397.00 90081.87 4133031.26 0.00 0.00 1.00 06:16:47 AM eth1 1396874.00 2793614.00 90032.99 4130385.46 0.00 0.00 0.00 06:16:48 AM eth1 1395842.00 2791600.00 89966.46 4127409.67 0.00 0.00 1.00 06:16:49 AM eth1 1395528.00 2791017.00 89946.17 4126551.24 0.00 0.00 0.00 06:16:50 AM eth1 1397891.00 2795716.00 90098.74 4133497.39 0.00 0.00 1.00 06:16:51 AM eth1 1394951.00 2789984.00 89908.96 4125022.51 0.00 0.00 0.00 06:16:52 AM eth1 1394608.00 2789190.00 89886.90 4123851.36 0.00 0.00 1.00 06:16:53 AM eth1 1395314.00 2790653.00 89934.33 4125983.09 0.00 0.00 0.00 06:16:54 AM eth1 1396115.00 2792276.00 89984.25 4128411.21 0.00 0.00 1.00 06:16:55 AM eth1 1396829.00 2793523.00 90030.19 4130250.28 0.00 0.00 0.00 Average: eth1 1396158.40 2792297.00 89987.09 4128439.35 0.00 0.00 0.50 lpaa23:~# tc -s -d qd sh dev eth1 | grep backlog backlog 7900052b 2609p requeues 173287 backlog 878120b 290p requeues 589 backlog 1068884b 354p requeues 5621 backlog 996212b 329p requeues 1088 backlog 984100b 325p requeues 115316 backlog 956848b 316p requeues 1781 backlog 1080996b 357p requeues 3047 backlog 975016b 322p requeues 24571 backlog 990156b 327p requeues 21274 (All 8 TX queues get a fair share of the traffic) 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 Small Queues (tcp_tsq_handler()) can hold one reference on sk->sk_wmem_alloc, preventing skb->ooo_okay being set. We should relax test done to set skb->ooo_okay to take care of this extra reference. Minimal truesize of skb containing one byte of payload is SKB_TRUESIZE(1) Without this fix, we have more chance locking flows into the wrong transmit queue. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 02 10月, 2014 1 次提交
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由 Eric Dumazet 提交于
Lets use a proper structure to clearly document and implement skb fast clones. Then, we might experiment more easily alternative layouts. This patch adds a new skb_fclone_busy() helper, used by tcp and xfrm, to stop leaking of implementation details. Signed-off-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 30 9月, 2014 1 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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- 29 9月, 2014 5 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 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 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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- 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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- 23 9月, 2014 1 次提交
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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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- 07 9月, 2014 1 次提交
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由 Neal Cardwell 提交于
The TCP_SKB_CB(skb)->when field no longer exists as of recent change 7faee5c0 ("tcp: remove TCP_SKB_CB(skb)->when"). And in any case, tcp_fragment() is called on already-transmitted packets from the __tcp_retransmit_skb() call site, so copying timestamps of any kind in this spot is quite sensible. Signed-off-by: NNeal Cardwell <ncardwell@google.com> Reported-by: NYuchung Cheng <ycheng@google.com> Acked-by: NEric Dumazet <edumazet@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 06 9月, 2014 1 次提交
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由 Eric Dumazet 提交于
After commit 740b0f18 ("tcp: switch rtt estimations to usec resolution"), we no longer need to maintain timestamps in two different fields. TCP_SKB_CB(skb)->when can be removed, as same information sits in skb_mstamp.stamp_jiffies 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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- 27 8月, 2014 1 次提交
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由 Christoph Lameter 提交于
Replace uses of get_cpu_var for address calculation through this_cpu_ptr. Cc: netdev@vger.kernel.org Cc: Eric Dumazet <edumazet@google.com> Acked-by: NDavid S. Miller <davem@davemloft.net> Signed-off-by: NChristoph Lameter <cl@linux.com> Signed-off-by: NTejun Heo <tj@kernel.org>
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- 15 8月, 2014 2 次提交
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由 Neal Cardwell 提交于
Make sure we use the correct address-family-specific function for handling MTU reductions from within tcp_release_cb(). Previously AF_INET6 sockets were incorrectly always using the IPv6 code path when sometimes they were handling IPv4 traffic and thus had an IPv4 dst. Signed-off-by: NNeal Cardwell <ncardwell@google.com> Signed-off-by: NEric Dumazet <edumazet@google.com> Diagnosed-by: NWillem de Bruijn <willemb@google.com> Fixes: 563d34d0 ("tcp: dont drop MTU reduction indications") Reviewed-by: NHannes Frederic Sowa <hannes@stressinduktion.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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由 Andrey Vagin 提交于
We don't know right timestamp for repaired skb-s. Wrong RTT estimations isn't good, because some congestion modules heavily depends on it. This patch adds the TCPCB_REPAIRED flag, which is included in TCPCB_RETRANS. Thanks to Eric for the advice how to fix this issue. This patch fixes the warning: [ 879.562947] WARNING: CPU: 0 PID: 2825 at net/ipv4/tcp_input.c:3078 tcp_ack+0x11f5/0x1380() [ 879.567253] CPU: 0 PID: 2825 Comm: socket-tcpbuf-l Not tainted 3.16.0-next-20140811 #1 [ 879.567829] Hardware name: Bochs Bochs, BIOS Bochs 01/01/2011 [ 879.568177] 0000000000000000 00000000c532680c ffff880039643d00 ffffffff817aa2d2 [ 879.568776] 0000000000000000 ffff880039643d38 ffffffff8109afbd ffff880039d6ba80 [ 879.569386] ffff88003a449800 000000002983d6bd 0000000000000000 000000002983d6bc [ 879.569982] Call Trace: [ 879.570264] [<ffffffff817aa2d2>] dump_stack+0x4d/0x66 [ 879.570599] [<ffffffff8109afbd>] warn_slowpath_common+0x7d/0xa0 [ 879.570935] [<ffffffff8109b0ea>] warn_slowpath_null+0x1a/0x20 [ 879.571292] [<ffffffff816d0a05>] tcp_ack+0x11f5/0x1380 [ 879.571614] [<ffffffff816d10bd>] tcp_rcv_established+0x1ed/0x710 [ 879.571958] [<ffffffff816dc9da>] tcp_v4_do_rcv+0x10a/0x370 [ 879.572315] [<ffffffff81657459>] release_sock+0x89/0x1d0 [ 879.572642] [<ffffffff816c81a0>] do_tcp_setsockopt.isra.36+0x120/0x860 [ 879.573000] [<ffffffff8110a52e>] ? rcu_read_lock_held+0x6e/0x80 [ 879.573352] [<ffffffff816c8912>] tcp_setsockopt+0x32/0x40 [ 879.573678] [<ffffffff81654ac4>] sock_common_setsockopt+0x14/0x20 [ 879.574031] [<ffffffff816537b0>] SyS_setsockopt+0x80/0xf0 [ 879.574393] [<ffffffff817b40a9>] system_call_fastpath+0x16/0x1b [ 879.574730] ---[ end trace a17cbc38eb8c5c00 ]--- v2: moving setting of skb->when for repaired skb-s in tcp_write_xmit, where it's set for other skb-s. Fixes: 431a9124 ("tcp: timestamp SYN+DATA messages") Fixes: 740b0f18 ("tcp: switch rtt estimations to usec resolution") Cc: Eric Dumazet <edumazet@google.com> Cc: Pavel Emelyanov <xemul@parallels.com> Cc: "David S. Miller" <davem@davemloft.net> Signed-off-by: NAndrey Vagin <avagin@openvz.org> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 14 8月, 2014 1 次提交
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由 Willem de Bruijn 提交于
Bytestream timestamps are correlated with a single byte in the skbuff, recorded in skb_shinfo(skb)->tskey. When fragmenting skbuffs, ensure that the tskey is set for the fragment in which the tskey falls (seqno <= tskey < end_seqno). The original implementation did not address fragmentation in tcp_fragment or tso_fragment. Add code to inspect the sequence numbers and move both tskey and the relevant tx_flags if necessary. Reported-by: NEric Dumazet <eric.dumazet@gmail.com> Signed-off-by: NWillem de Bruijn <willemb@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 16 7月, 2014 1 次提交
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由 Christoph Paasch 提交于
Since Yuchung's 9b44190d (tcp: refactor F-RTO), tcp_enter_cwr is always called with set_ssthresh = 1. Thus, we can remove this argument from tcp_enter_cwr. Further, as we remove this one, tcp_init_cwnd_reduction is then always called with set_ssthresh = true, and so we can get rid of this argument as well. Cc: Yuchung Cheng <ycheng@google.com> Signed-off-by: NChristoph Paasch <christoph.paasch@uclouvain.be> Acked-by: NYuchung Cheng <ycheng@google.com> Signed-off-by: NDavid S. Miller <davem@davemloft.net>
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- 08 7月, 2014 1 次提交
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由 Yuchung Cheng 提交于
The undo code assumes that, upon entering loss recovery, TCP 1) always retransmit something 2) the retransmission never fails locally (e.g., qdisc drop) so undo_marker is set in tcp_enter_recovery() and undo_retrans is incremented only when tcp_retransmit_skb() is successful. When the assumption is broken because TCP's cwnd is too small to retransmit or the retransmit fails locally. The next (DUP)ACK would incorrectly revert the cwnd and the congestion state in tcp_try_undo_dsack() or tcp_may_undo(). Subsequent (DUP)ACKs may enter the recovery state. The sender repeatedly enter and (incorrectly) exit recovery states if the retransmits continue to fail locally while receiving (DUP)ACKs. The fix is to initialize undo_retrans to -1 and start counting on the first retransmission. Always increment undo_retrans even if the retransmissions fail locally because they couldn't cause DSACKs to undo the cwnd reduction. 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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