1. 08 2月, 2015 1 次提交
    • N
      tcp: helpers to mitigate ACK loops by rate-limiting out-of-window dupacks · 032ee423
      Neal Cardwell 提交于
      Helpers for mitigating ACK loops by rate-limiting dupacks sent in
      response to incoming out-of-window packets.
      
      This patch includes:
      
      - rate-limiting logic
      - sysctl to control how often we allow dupacks to out-of-window packets
      - SNMP counter for cases where we rate-limited our dupack sending
      
      The rate-limiting logic in this patch decides to not send dupacks in
      response to out-of-window segments if (a) they are SYNs or pure ACKs
      and (b) the remote endpoint is sending them faster than the configured
      rate limit.
      
      We rate-limit our responses rather than blocking them entirely or
      resetting the connection, because legitimate connections can rely on
      dupacks in response to some out-of-window segments. For example, zero
      window probes are typically sent with a sequence number that is below
      the current window, and ZWPs thus expect to thus elicit a dupack in
      response.
      
      We allow dupacks in response to TCP segments with data, because these
      may be spurious retransmissions for which the remote endpoint wants to
      receive DSACKs. This is safe because segments with data can't
      realistically be part of ACK loops, which by their nature consist of
      each side sending pure/data-less ACKs to each other.
      
      The dupack interval is controlled by a new sysctl knob,
      tcp_invalid_ratelimit, given in milliseconds, in case an administrator
      needs to dial this upward in the face of a high-rate DoS attack. The
      name and units are chosen to be analogous to the existing analogous
      knob for ICMP, icmp_ratelimit.
      
      The default value for tcp_invalid_ratelimit is 500ms, which allows at
      most one such dupack per 500ms. This is chosen to be 2x faster than
      the 1-second minimum RTO interval allowed by RFC 6298 (section 2, rule
      2.4). We allow the extra 2x factor because network delay variations
      can cause packets sent at 1 second intervals to be compressed and
      arrive much closer.
      Reported-by: NAvery Fay <avery@mixpanel.com>
      Signed-off-by: NNeal Cardwell <ncardwell@google.com>
      Signed-off-by: NYuchung Cheng <ycheng@google.com>
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      032ee423
  2. 05 2月, 2015 1 次提交
    • E
      tcp: do not pace pure ack packets · 98781965
      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>
      98781965
  3. 29 1月, 2015 1 次提交
    • N
      tcp: stretch ACK fixes prep · e73ebb08
      Neal Cardwell 提交于
      LRO, GRO, delayed ACKs, and middleboxes can cause "stretch ACKs" that
      cover more than the RFC-specified maximum of 2 packets. These stretch
      ACKs can cause serious performance shortfalls in common congestion
      control algorithms that were designed and tuned years ago with
      receiver hosts that were not using LRO or GRO, and were instead
      politely ACKing every other packet.
      
      This patch series fixes Reno and CUBIC to handle stretch ACKs.
      
      This patch prepares for the upcoming stretch ACK bug fix patches. It
      adds an "acked" parameter to tcp_cong_avoid_ai() to allow for future
      fixes to tcp_cong_avoid_ai() to correctly handle stretch ACKs, and
      changes all congestion control algorithms to pass in 1 for the ACKed
      count. It also changes tcp_slow_start() to return the number of packet
      ACK "credits" that were not processed in slow start mode, and can be
      processed by the congestion control module in additive increase mode.
      
      In future patches we will fix tcp_cong_avoid_ai() to handle stretch
      ACKs, and fix Reno and CUBIC handling of stretch ACKs in slow start
      and additive increase mode.
      Reported-by: NEyal Perry <eyalpe@mellanox.com>
      Signed-off-by: NNeal Cardwell <ncardwell@google.com>
      Signed-off-by: NYuchung Cheng <ycheng@google.com>
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      e73ebb08
  4. 06 1月, 2015 3 次提交
    • D
      net: tcp: add per route congestion control · 81164413
      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>
      81164413
    • D
      net: tcp: add RTAX_CC_ALGO fib handling · ea697639
      Daniel Borkmann 提交于
      This patch adds the minimum necessary for the RTAX_CC_ALGO congestion
      control metric to be set up and dumped back to user space.
      
      While the internal representation of RTAX_CC_ALGO is handled as a u32
      key, we avoided to expose this implementation detail to user space, thus
      instead, we chose the netlink attribute that is being exchanged between
      user space to be the actual congestion control algorithm name, similarly
      as in the setsockopt(2) API in order to allow for maximum flexibility,
      even for 3rd party modules.
      
      It is a bit unfortunate that RTAX_QUICKACK used up a whole RTAX slot as
      it should have been stored in RTAX_FEATURES instead, we first thought
      about reusing it for the congestion control key, but it brings more
      complications and/or confusion than worth it.
      
      Joint work with Florian Westphal.
      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>
      ea697639
    • D
      net: tcp: add key management to congestion control · c5c6a8ab
      Daniel Borkmann 提交于
      This patch adds necessary infrastructure to the congestion control
      framework for later per route congestion control support.
      
      For a per route congestion control possibility, our aim is to store
      a unique u32 key identifier into dst metrics, which can then be
      mapped into a tcp_congestion_ops struct. We argue that having a
      RTAX key entry is the most simple, generic and easy way to manage,
      and also keeps the memory footprint of dst entries lower on 64 bit
      than with storing a pointer directly, for example. Having a unique
      key id also allows for decoupling actual TCP congestion control
      module management from the FIB layer, i.e. we don't have to care
      about expensive module refcounting inside the FIB at this point.
      
      We first thought of using an IDR store for the realization, which
      takes over dynamic assignment of unused key space and also performs
      the key to pointer mapping in RCU. While doing so, we stumbled upon
      the issue that due to the nature of dynamic key distribution, it
      just so happens, arguably in very rare occasions, that excessive
      module loads and unloads can lead to a possible reuse of previously
      used key space. Thus, previously stale keys in the dst metric are
      now being reassigned to a different congestion control algorithm,
      which might lead to unexpected behaviour. One way to resolve this
      would have been to walk FIBs on the actually rare occasion of a
      module unload and reset the metric keys for each FIB in each netns,
      but that's just very costly.
      
      Therefore, we argue a better solution is to reuse the unique
      congestion control algorithm name member and map that into u32 key
      space through jhash. For that, we split the flags attribute (as it
      currently uses 2 bits only anyway) into two u32 attributes, flags
      and key, so that we can keep the cacheline boundary of 2 cachelines
      on x86_64 and cache the precalculated key at registration time for
      the fast path. On average we might expect 2 - 4 modules being loaded
      worst case perhaps 15, so a key collision possibility is extremely
      low, and guaranteed collision-free on LE/BE for all in-tree modules.
      Overall this results in much simpler code, and all without the
      overhead of an IDR. Due to the deterministic nature, modules can
      now be unloaded, the congestion control algorithm for a specific
      but unloaded key will fall back to the default one, and on module
      reload time it will switch back to the expected algorithm
      transparently.
      
      Joint work with Florian Westphal.
      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>
      c5c6a8ab
  5. 05 11月, 2014 2 次提交
    • F
      net: allow setting ecn via routing table · f7b3bec6
      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>
      f7b3bec6
    • F
      syncookies: split cookie_check_timestamp() into two functions · f1673381
      Florian Westphal 提交于
      The function cookie_check_timestamp(), both called from IPv4/6 context,
      is being used to decode the echoed timestamp from the SYN/ACK into TCP
      options used for follow-up communication with the peer.
      
      We can remove ECN handling from that function, split it into a separate
      one, and simply rename the original function into cookie_decode_options().
      cookie_decode_options() just fills in tcp_option struct based on the
      echoed timestamp received from the peer. Anything that fails in this
      function will actually discard the request socket.
      
      While this is the natural place for decoding options such as ECN which
      commit 172d69e6 ("syncookies: add support for ECN") added, we argue
      that in particular for ECN handling, it can be checked at a later point
      in time as the request sock would actually not need to be dropped from
      this, but just ECN support turned off.
      
      Therefore, we split this functionality into cookie_ecn_ok(), which tells
      us if the timestamp indicates ECN support AND the tcp_ecn sysctl is enabled.
      
      This prepares for per-route ECN support: just looking at the tcp_ecn sysctl
      won't be enough anymore at that point; if the timestamp indicates ECN
      and sysctl tcp_ecn == 0, we will also need to check the ECN dst metric.
      
      This would mean adding a route lookup to cookie_check_timestamp(), which
      we definitely want to avoid. As we already do a route lookup at a later
      point in cookie_{v4,v6}_check(), we can simply make use of that as well
      for the new cookie_ecn_ok() function w/o any additional cost.
      
      Joint work with Daniel Borkmann.
      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>
      f1673381
  6. 30 10月, 2014 1 次提交
    • E
      tcp: allow for bigger reordering level · dca145ff
      Eric Dumazet 提交于
      While testing upcoming Yaogong patch (converting out of order queue
      into an RB tree), I hit the max reordering level of linux TCP stack.
      
      Reordering level was limited to 127 for no good reason, and some
      network setups [1] can easily reach this limit and get limited
      throughput.
      
      Allow a new max limit of 300, and add a sysctl to allow admins to even
      allow bigger (or lower) values if needed.
      
      [1] Aggregation of links, per packet load balancing, fabrics not doing
       deep packet inspections, alternative TCP congestion modules...
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Cc: Yaogong Wang <wygivan@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      dca145ff
  7. 24 10月, 2014 1 次提交
  8. 19 10月, 2014 1 次提交
  9. 18 10月, 2014 3 次提交
  10. 01 10月, 2014 1 次提交
  11. 30 9月, 2014 1 次提交
  12. 29 9月, 2014 6 次提交
    • F
      net: tcp: more detailed ACK events and events for CE marked packets · 9890092e
      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>
      9890092e
    • F
      net: tcp: split ack slow/fast events from cwnd_event · 7354c8c3
      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>
      7354c8c3
    • D
      net: tcp: add flag for ca to indicate that ECN is required · 30e502a3
      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>
      30e502a3
    • F
      net: tcp: assign tcp cong_ops when tcp sk is created · 55d8694f
      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>
      55d8694f
    • E
      tcp: change tcp_skb_pcount() location · cd7d8498
      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>
      cd7d8498
    • E
      tcp: better TCP_SKB_CB layout to reduce cache line misses · 971f10ec
      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>
      971f10ec
  13. 28 9月, 2014 2 次提交
  14. 06 9月, 2014 2 次提交
  15. 15 8月, 2014 3 次提交
    • H
      tcp: don't allow syn packets without timestamps to pass tcp_tw_recycle logic · a26552af
      Hannes Frederic Sowa 提交于
      tcp_tw_recycle heavily relies on tcp timestamps to build a per-host
      ordering of incoming connections and teardowns without the need to
      hold state on a specific quadruple for TCP_TIMEWAIT_LEN, but only for
      the last measured RTO. To do so, we keep the last seen timestamp in a
      per-host indexed data structure and verify if the incoming timestamp
      in a connection request is strictly greater than the saved one during
      last connection teardown. Thus we can verify later on that no old data
      packets will be accepted by the new connection.
      
      During moving a socket to time-wait state we already verify if timestamps
      where seen on a connection. Only if that was the case we let the
      time-wait socket expire after the RTO, otherwise normal TCP_TIMEWAIT_LEN
      will be used. But we don't verify this on incoming SYN packets. If a
      connection teardown was less than TCP_PAWS_MSL seconds in the past we
      cannot guarantee to not accept data packets from an old connection if
      no timestamps are present. We should drop this SYN packet. This patch
      closes this loophole.
      
      Please note, this patch does not make tcp_tw_recycle in any way more
      usable but only adds another safety check:
      Sporadic drops of SYN packets because of reordering in the network or
      in the socket backlog queues can happen. Users behing NAT trying to
      connect to a tcp_tw_recycle enabled server can get caught in blackholes
      and their connection requests may regullary get dropped because hosts
      behind an address translator don't have synchronized tcp timestamp clocks.
      tcp_tw_recycle cannot work if peers don't have tcp timestamps enabled.
      
      In general, use of tcp_tw_recycle is disadvised.
      
      Cc: Eric Dumazet <eric.dumazet@gmail.com>
      Cc: Florian Westphal <fw@strlen.de>
      Signed-off-by: NHannes Frederic Sowa <hannes@stressinduktion.org>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      a26552af
    • N
      tcp: fix tcp_release_cb() to dispatch via address family for mtu_reduced() · 4fab9071
      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>
      4fab9071
    • A
      tcp: don't use timestamp from repaired skb-s to calculate RTT (v2) · 9d186cac
      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>
      9d186cac
  16. 06 8月, 2014 1 次提交
    • N
      tcp: reduce spurious retransmits due to transient SACK reneging · 5ae344c9
      Neal Cardwell 提交于
      This commit reduces spurious retransmits due to apparent SACK reneging
      by only reacting to SACK reneging that persists for a short delay.
      
      When a sequence space hole at snd_una is filled, some TCP receivers
      send a series of ACKs as they apparently scan their out-of-order queue
      and cumulatively ACK all the packets that have now been consecutiveyly
      received. This is essentially misbehavior B in "Misbehaviors in TCP
      SACK generation" ACM SIGCOMM Computer Communication Review, April
      2011, so we suspect that this is from several common OSes (Windows
      2000, Windows Server 2003, Windows XP). However, this issue has also
      been seen in other cases, e.g. the netdev thread "TCP being hoodwinked
      into spurious retransmissions by lack of timestamps?" from March 2014,
      where the receiver was thought to be a BSD box.
      
      Since snd_una would temporarily be adjacent to a previously SACKed
      range in these scenarios, this receiver behavior triggered the Linux
      SACK reneging code path in the sender. This led the sender to clear
      the SACK scoreboard, enter CA_Loss, and spuriously retransmit
      (potentially) every packet from the entire write queue at line rate
      just a few milliseconds before the ACK for each packet arrives at the
      sender.
      
      To avoid such situations, now when a sender sees apparent reneging it
      does not yet retransmit, but rather adjusts the RTO timer to give the
      receiver a little time (max(RTT/2, 10ms)) to send us some more ACKs
      that will restore sanity to the SACK scoreboard. If the reneging
      persists until this RTO then, as before, we clear the SACK scoreboard
      and enter CA_Loss.
      
      A 10ms delay tolerates a receiver sending such a stream of ACKs at
      56Kbit/sec. And to allow for receivers with slower or more congested
      paths, we wait for at least RTT/2.
      
      We validated the resulting max(RTT/2, 10ms) delay formula with a mix
      of North American and South American Google web server traffic, and
      found that for ACKs displaying transient reneging:
      
       (1) 90% of inter-ACK delays were less than 10ms
       (2) 99% of inter-ACK delays were less than RTT/2
      
      In tests on Google web servers this commit reduced reneging events by
      75%-90% (as measured by the TcpExtTCPSACKReneging counter), without
      any measurable impact on latency for user HTTP and SPDY requests.
      Signed-off-by: NNeal Cardwell <ncardwell@google.com>
      Signed-off-by: NYuchung Cheng <ycheng@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      5ae344c9
  17. 16 7月, 2014 1 次提交
  18. 08 7月, 2014 1 次提交
    • N
      tcp: switch snt_synack back to measuring transmit time of first SYNACK · 86c6a2c7
      Neal Cardwell 提交于
      Always store in snt_synack the time at which the server received the
      first client SYN and attempted to send the first SYNACK.
      
      Recent commit aa27fc50 ("tcp: tcp_v[46]_conn_request: fix snt_synack
      initialization") resolved an inconsistency between IPv4 and IPv6 in
      the initialization of snt_synack. This commit brings back the idea
      from 843f4a55 (tcp: use tcp_v4_send_synack on first SYN-ACK), which
      was going for the original behavior of snt_synack from the commit
      where it was added in 9ad7c049 ("tcp: RFC2988bis + taking RTT
      sample from 3WHS for the passive open side") in v3.1.
      
      In addition to being simpler (and probably a tiny bit faster),
      unconditionally storing the time of the first SYNACK attempt has been
      useful because it allows calculating a performance metric quantifying
      how long it took to establish a passive TCP connection.
      Signed-off-by: NNeal Cardwell <ncardwell@google.com>
      Signed-off-by: NYuchung Cheng <ycheng@google.com>
      Cc: Octavian Purdila <octavian.purdila@intel.com>
      Cc: Jerry Chu <hkchu@google.com>
      Acked-by: NOctavian Purdila <octavian.purdila@intel.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      86c6a2c7
  19. 28 6月, 2014 8 次提交