1. 27 7月, 2012 1 次提交
  2. 23 7月, 2012 1 次提交
    • E
      tcp: dont drop MTU reduction indications · 563d34d0
      Eric Dumazet 提交于
      ICMP messages generated in output path if frame length is bigger than
      mtu are actually lost because socket is owned by user (doing the xmit)
      
      One example is the ipgre_tunnel_xmit() calling
      icmp_send(skb, ICMP_DEST_UNREACH, ICMP_FRAG_NEEDED, htonl(mtu));
      
      We had a similar case fixed in commit a34a101e (ipv6: disable GSO on
      sockets hitting dst_allfrag).
      
      Problem of such fix is that it relied on retransmit timers, so short tcp
      sessions paid a too big latency increase price.
      
      This patch uses the tcp_release_cb() infrastructure so that MTU
      reduction messages (ICMP messages) are not lost, and no extra delay
      is added in TCP transmits.
      Reported-by: NMaciej Żenczykowski <maze@google.com>
      Diagnosed-by: NNeal Cardwell <ncardwell@google.com>
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Cc: Nandita Dukkipati <nanditad@google.com>
      Cc: Tom Herbert <therbert@google.com>
      Cc: Tore Anderson <tore@fud.no>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      563d34d0
  3. 20 7月, 2012 1 次提交
    • Y
      net-tcp: Fast Open base · 2100c8d2
      Yuchung Cheng 提交于
      This patch impelements the common code for both the client and server.
      
      1. TCP Fast Open option processing. Since Fast Open does not have an
         option number assigned by IANA yet, it shares the experiment option
         code 254 by implementing draft-ietf-tcpm-experimental-options
         with a 16 bits magic number 0xF989. This enables global experiments
         without clashing the scarce(2) experimental options available for TCP.
      
         When the draft status becomes standard (maybe), the client should
         switch to the new option number assigned while the server supports
         both numbers for transistion.
      
      2. The new sysctl tcp_fastopen
      
      3. A place holder init function
      Signed-off-by: NYuchung Cheng <ycheng@google.com>
      Acked-by: NEric Dumazet <edumazet@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      2100c8d2
  4. 17 7月, 2012 1 次提交
    • D
      net: Pass optional SKB and SK arguments to dst_ops->{update_pmtu,redirect}() · 6700c270
      David S. Miller 提交于
      This will be used so that we can compose a full flow key.
      
      Even though we have a route in this context, we need more.  In the
      future the routes will be without destination address, source address,
      etc. keying.  One ipv4 route will cover entire subnets, etc.
      
      In this environment we have to have a way to possess persistent storage
      for redirects and PMTU information.  This persistent storage will exist
      in the FIB tables, and that's why we'll need to be able to rebuild a
      full lookup flow key here.  Using that flow key will do a fib_lookup()
      and create/update the persistent entry.
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      6700c270
  5. 16 7月, 2012 1 次提交
  6. 12 7月, 2012 3 次提交
    • D
      net: Remove checks for dst_ops->redirect being NULL. · 1ed5c48f
      David S. Miller 提交于
      No longer necessary.
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      1ed5c48f
    • D
    • E
      tcp: TCP Small Queues · 46d3ceab
      Eric Dumazet 提交于
      This introduce TSQ (TCP Small Queues)
      
      TSQ goal is to reduce number of TCP packets in xmit queues (qdisc &
      device queues), to reduce RTT and cwnd bias, part of the bufferbloat
      problem.
      
      sk->sk_wmem_alloc not allowed to grow above a given limit,
      allowing no more than ~128KB [1] per tcp socket in qdisc/dev layers at a
      given time.
      
      TSO packets are sized/capped to half the limit, so that we have two
      TSO packets in flight, allowing better bandwidth use.
      
      As a side effect, setting the limit to 40000 automatically reduces the
      standard gso max limit (65536) to 40000/2 : It can help to reduce
      latencies of high prio packets, having smaller TSO packets.
      
      This means we divert sock_wfree() to a tcp_wfree() handler, to
      queue/send following frames when skb_orphan() [2] is called for the
      already queued skbs.
      
      Results on my dev machines (tg3/ixgbe nics) are really impressive,
      using standard pfifo_fast, and with or without TSO/GSO.
      
      Without reduction of nominal bandwidth, we have reduction of buffering
      per bulk sender :
      < 1ms on Gbit (instead of 50ms with TSO)
      < 8ms on 100Mbit (instead of 132 ms)
      
      I no longer have 4 MBytes backlogged in qdisc by a single netperf
      session, and both side socket autotuning no longer use 4 Mbytes.
      
      As skb destructor cannot restart xmit itself ( as qdisc lock might be
      taken at this point ), we delegate the work to a tasklet. We use one
      tasklest per cpu for performance reasons.
      
      If tasklet finds a socket owned by the user, it sets TSQ_OWNED flag.
      This flag is tested in a new protocol method called from release_sock(),
      to eventually send new segments.
      
      [1] New /proc/sys/net/ipv4/tcp_limit_output_bytes tunable
      [2] skb_orphan() is usually called at TX completion time,
        but some drivers call it in their start_xmit() handler.
        These drivers should at least use BQL, or else a single TCP
        session can still fill the whole NIC TX ring, since TSQ will
        have no effect.
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Cc: Dave Taht <dave.taht@bufferbloat.net>
      Cc: Tom Herbert <therbert@google.com>
      Cc: Matt Mathis <mattmathis@google.com>
      Cc: Yuchung Cheng <ycheng@google.com>
      Cc: Nandita Dukkipati <nanditad@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      46d3ceab
  7. 11 7月, 2012 3 次提交
  8. 05 7月, 2012 1 次提交
  9. 29 6月, 2012 3 次提交
  10. 26 6月, 2012 1 次提交
  11. 16 6月, 2012 1 次提交
    • D
      ipv6: Handle PMTU in ICMP error handlers. · 81aded24
      David S. Miller 提交于
      One tricky issue on the ipv6 side vs. ipv4 is that the ICMP callouts
      to handle the error pass the 32-bit info cookie in network byte order
      whereas ipv4 passes it around in host byte order.
      
      Like the ipv4 side, we have two helper functions.  One for when we
      have a socket context and one for when we do not.
      
      ip6ip6 tunnels are not handled here, because they handle PMTU events
      by essentially relaying another ICMP packet-too-big message back to
      the original sender.
      
      This patch allows us to get rid of rt6_do_pmtu_disc().  It handles all
      kinds of situations that simply cannot happen when we do the PMTU
      update directly using a fully resolved route.
      
      In fact, the "plen == 128" check in ip6_rt_update_pmtu() can very
      likely be removed or changed into a BUG_ON() check.  We should never
      have a prefixed ipv6 route when we get there.
      
      Another piece of strange history here is that TCP and DCCP, unlike in
      ipv4, never invoke the update_pmtu() method from their ICMP error
      handlers.  This is incredibly astonishing since this is the context
      where we have the most accurate context in which to make a PMTU
      update, namely we have a fully connected socket and associated cached
      socket route.
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      81aded24
  12. 10 6月, 2012 1 次提交
  13. 09 6月, 2012 3 次提交
    • D
      tcp: Get rid of inetpeer special cases. · 4670fd81
      David S. Miller 提交于
      The get_peer method TCP uses is full of special cases that make no
      sense accommodating, and it also gets in the way of doing more
      reasonable things here.
      
      First of all, if the socket doesn't have a usable cached route, there
      is no sense in trying to optimize timewait recycling.
      
      Likewise for the case where we have IP options, such as SRR enabled,
      that make the IP header destination address (and thus the destination
      address of the route key) differ from that of the connection's
      destination address.
      
      Just return a NULL peer in these cases, and thus we're also able to
      get rid of the clumsy inetpeer release logic.
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      4670fd81
    • D
      inet: Create and use rt{,6}_get_peer_create(). · fbfe95a4
      David S. Miller 提交于
      There's a lot of places that open-code rt{,6}_get_peer() only because
      they want to set 'create' to one.  So add an rt{,6}_get_peer_create()
      for their sake.
      
      There were also a few spots open-coding plain rt{,6}_get_peer() and
      those are transformed here as well.
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      fbfe95a4
    • G
      inetpeer: add parameter net for inet_getpeer_v4,v6 · 54db0cc2
      Gao feng 提交于
      add struct net as a parameter of inet_getpeer_v[4,6],
      use net to replace &init_net.
      
      and modify some places to provide net for inet_getpeer_v[4,6]
      Signed-off-by: NGao feng <gaofeng@cn.fujitsu.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      54db0cc2
  14. 04 6月, 2012 1 次提交
  15. 02 6月, 2012 1 次提交
    • E
      tcp: reflect SYN queue_mapping into SYNACK packets · fff32699
      Eric Dumazet 提交于
      While testing how linux behaves on SYNFLOOD attack on multiqueue device
      (ixgbe), I found that SYNACK messages were dropped at Qdisc level
      because we send them all on a single queue.
      
      Obvious choice is to reflect incoming SYN packet @queue_mapping to
      SYNACK packet.
      
      Under stress, my machine could only send 25.000 SYNACK per second (for
      200.000 incoming SYN per second). NIC : ixgbe with 16 rx/tx queues.
      
      After patch, not a single SYNACK is dropped.
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Cc: Hans Schillstrom <hans.schillstrom@ericsson.com>
      Cc: Jesper Dangaard Brouer <brouer@redhat.com>
      Cc: Neal Cardwell <ncardwell@google.com>
      Cc: Tom Herbert <therbert@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      fff32699
  16. 18 5月, 2012 1 次提交
  17. 16 5月, 2012 1 次提交
  18. 05 5月, 2012 1 次提交
    • E
      tcp: be more strict before accepting ECN negociation · bd14b1b2
      Eric Dumazet 提交于
      It appears some networks play bad games with the two bits reserved for
      ECN. This can trigger false congestion notifications and very slow
      transferts.
      
      Since RFC 3168 (6.1.1) forbids SYN packets to carry CT bits, we can
      disable TCP ECN negociation if it happens we receive mangled CT bits in
      the SYN packet.
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Cc: Perry Lorier <perryl@google.com>
      Cc: Matt Mathis <mattmathis@google.com>
      Cc: Yuchung Cheng <ycheng@google.com>
      Cc: Neal Cardwell <ncardwell@google.com>
      Cc: Wilmer van der Gaast <wilmer@google.com>
      Cc: Ankur Jain <jankur@google.com>
      Cc: Tom Herbert <therbert@google.com>
      Cc: Dave Täht <dave.taht@bufferbloat.net>
      Acked-by: NNeal Cardwell <ncardwell@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      bd14b1b2
  19. 27 4月, 2012 1 次提交
    • E
      ipv6: RTAX_FEATURE_ALLFRAG causes inefficient TCP segment sizing · 67469601
      Eric Dumazet 提交于
      Quoting Tore Anderson from :
      https://bugzilla.kernel.org/show_bug.cgi?id=42572
      
      When RTAX_FEATURE_ALLFRAG is set on a route, the effective TCP segment
      size does not take into account the size of the IPv6 Fragmentation
      header that needs to be included in outbound packets, causing every
      transmitted TCP segment to be fragmented across two IPv6 packets, the
      latter of which will only contain 8 bytes of actual payload.
      
      RTAX_FEATURE_ALLFRAG is typically set on a route in response to
      receving a ICMPv6 Packet Too Big message indicating a Path MTU of less
      than 1280 bytes. 1280 bytes is the minimum IPv6 MTU, however ICMPv6
      PTBs with MTU < 1280 are still valid, in particular when an IPv6
      packet is sent to an IPv4 destination through a stateless translator.
      Any ICMPv4 Need To Fragment packets originated from the IPv4 part of
      the path will be translated to ICMPv6 PTB which may then indicate an
      MTU of less than 1280.
      
      The Linux kernel refuses to reduce the effective MTU to anything below
      1280 bytes, instead it sets it to exactly 1280 bytes, and
      RTAX_FEATURE_ALLFRAG is also set. However, the TCP segment size appears
      to be set to 1240 bytes (1280 Path MTU - 40 bytes of IPv6 header),
      instead of 1232 (additionally taking into account the 8 bytes required
      by the IPv6 Fragmentation extension header).
      
      This in turn results in rather inefficient transmission, as every
      transmitted TCP segment now is split in two fragments containing
      1232+8 bytes of payload.
      
      After this patch, all the outgoing packets that includes a
      Fragmentation header all are "atomic" or "non-fragmented" fragments,
      i.e., they both have Offset=0 and More Fragments=0.
      
      With help from David S. Miller
      Reported-by: NTore Anderson <tore@fud.no>
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Cc: Maciej Żenczykowski <maze@google.com>
      Cc: Tom Herbert <therbert@google.com>
      Tested-by: NTore Anderson <tore@fud.no>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      67469601
  20. 24 4月, 2012 2 次提交
    • E
      tcp: sk_add_backlog() is too agressive for TCP · da882c1f
      Eric Dumazet 提交于
      While investigating TCP performance problems on 10Gb+ links, we found a
      tcp sender was dropping lot of incoming ACKS because of sk_rcvbuf limit
      in sk_add_backlog(), especially if receiver doesnt use GRO/LRO and sends
      one ACK every two MSS segments.
      
      A sender usually tweaks sk_sndbuf, but sk_rcvbuf stays at its default
      value (87380), allowing a too small backlog.
      
      A TCP ACK, even being small, can consume nearly same truesize space than
      outgoing packets. Using sk_rcvbuf + sk_sndbuf as a limit makes sense and
      is fast to compute.
      
      Performance results on netperf, single flow, receiver with disabled
      GRO/LRO : 7500 Mbits instead of 6050 Mbits, no more TCPBacklogDrop
      increments at sender.
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Cc: Neal Cardwell <ncardwell@google.com>
      Cc: Tom Herbert <therbert@google.com>
      Cc: Maciej Żenczykowski <maze@google.com>
      Cc: Yuchung Cheng <ycheng@google.com>
      Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
      Cc: Rick Jones <rick.jones2@hp.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      da882c1f
    • E
      net: add a limit parameter to sk_add_backlog() · f545a38f
      Eric Dumazet 提交于
      sk_add_backlog() & sk_rcvqueues_full() hard coded sk_rcvbuf as the
      memory limit. We need to make this limit a parameter for TCP use.
      
      No functional change expected in this patch, all callers still using the
      old sk_rcvbuf limit.
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Cc: Neal Cardwell <ncardwell@google.com>
      Cc: Tom Herbert <therbert@google.com>
      Cc: Maciej Żenczykowski <maze@google.com>
      Cc: Yuchung Cheng <ycheng@google.com>
      Cc: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
      Cc: Rick Jones <rick.jones2@hp.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      f545a38f
  21. 23 4月, 2012 2 次提交
  22. 22 4月, 2012 1 次提交
    • N
      tcp: move duplicate code from tcp_v4_init_sock()/tcp_v6_init_sock() · 900f65d3
      Neal Cardwell 提交于
      This commit moves the (substantial) common code shared between
      tcp_v4_init_sock() and tcp_v6_init_sock() to a new address-family
      independent function, tcp_init_sock().
      
      Centralizing this functionality should help avoid drift issues,
      e.g. where the IPv4 side is updated without a corresponding update to
      IPv6. There was already some drift: IPv4 initialized snd_cwnd to
      TCP_INIT_CWND, while the IPv6 side was still initializing snd_cwnd to
      2 (in this case it should not matter, since snd_cwnd is also
      initialized in tcp_init_metrics(), but the general risks and
      maintenance overhead remain).
      
      When diffing the old and new code, note that new tcp_init_sock()
      function uses the order of steps from the tcp_v4_init_sock()
      implementation (the order is slightly different in
      tcp_v6_init_sock()).
      Signed-off-by: NNeal Cardwell <ncardwell@google.com>
      Acked-by: NEric Dumazet <edumazet@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      900f65d3
  23. 20 4月, 2012 1 次提交
  24. 06 4月, 2012 1 次提交
  25. 13 2月, 2012 1 次提交
    • J
      net: implement IP_RECVTOS for IP_PKTOPTIONS · 4c507d28
      Jiri Benc 提交于
      Currently, it is not easily possible to get TOS/DSCP value of packets from
      an incoming TCP stream. The mechanism is there, IP_PKTOPTIONS getsockopt
      with IP_RECVTOS set, the same way as incoming TTL can be queried. This is
      not actually implemented for TOS, though.
      
      This patch adds this functionality, both for IPv4 (IP_PKTOPTIONS) and IPv6
      (IPV6_2292PKTOPTIONS). For IPv4, like in the IP_RECVTTL case, the value of
      the TOS field is stored from the other party's ACK.
      
      This is needed for proxies which require DSCP transparency. One such example
      is at http://zph.bratcheda.org/.
      Signed-off-by: NJiri Benc <jbenc@redhat.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      4c507d28
  26. 02 2月, 2012 1 次提交
    • S
      tcp: md5: RST: getting md5 key from listener · 658ddaaf
      Shawn Lu 提交于
      TCP RST mechanism is broken in TCP md5(RFC2385). When
      connection is gone, md5 key is lost, sending RST
      without md5 hash is deem to ignored by peer. This can
      be a problem since RST help protocal like bgp to fast
      recove from peer crash.
      
      In most case, users of tcp md5, such as bgp and ldp,
      have listener on both sides to accept connection from peer.
      md5 keys for peers are saved in listening socket.
      
      There are two cases in finding md5 key when connection is
      lost:
      1.Passive receive RST: The message is send to well known port,
      tcp will associate it with listner. md5 key is gotten from
      listener.
      
      2.Active receive RST (no sock): The message is send to ative
      side, there is no socket associated with the message. In this
      case, finding listener from source port, then find md5 key from
      listener.
      
      we are not loosing sercuriy here:
      packet is checked with md5 hash. No RST is generated
      if md5 hash doesn't match or no md5 key can be found.
      Signed-off-by: NShawn Lu <shawn.lu@ericsson.com>
      Signed-off-by: NEric Dumazet <eric.dumazet@gmail.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      658ddaaf
  27. 01 2月, 2012 3 次提交
  28. 23 1月, 2012 1 次提交