1. 14 1月, 2015 1 次提交
  2. 13 1月, 2015 1 次提交
  3. 12 1月, 2015 1 次提交
    • W
      doc: fix the compile fix of txtimestamp.c · d3b4b261
      Willem de Bruijn 提交于
      A fix to ipv6 structure definitions removed the now superfluous
      definition of in6_pktinfo in this file.
      
      But, use of the glibc definition requires defining _GNU_SOURCE
      (see also https://sourceware.org/bugzilla/show_bug.cgi?id=6775).
      
      Before this change, the following would fail for me:
      
        make
        make headers_install
        make M=Documentation/networking/timestamping
      
      with
      
        Documentation/networking/timestamping/txtimestamp.c: In function '__recv_errmsg_cmsg':
        Documentation/networking/timestamping/txtimestamp.c:205:33: error: dereferencing pointer to incomplete type
        Documentation/networking/timestamping/txtimestamp.c:206:23: error: dereferencing pointer to incomplete type
      
      After this patch compilation succeeded.
      
      Fixes: cd91cc5b ("doc: fix the compile error of txtimestamp.c")
      Signed-off-by: NWillem de Bruijn <willemb@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      d3b4b261
  4. 09 1月, 2015 1 次提交
  5. 23 12月, 2014 1 次提交
  6. 16 12月, 2014 1 次提交
  7. 10 12月, 2014 1 次提交
  8. 09 12月, 2014 2 次提交
    • W
      net-timestamp: expand documentation and test · cbd3aad5
      Willem de Bruijn 提交于
      Documentation:
        expand explanation of timestamp counter
      
      Test:
        new: flag -I requests and prints PKTINFO
        new: flag -x prints payload (possibly truncated)
        fix: remove pretty print that breaks common flag '-l 1'
      Signed-off-by: NWillem de Bruijn <willemb@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      cbd3aad5
    • W
      net-timestamp: allow reading recv cmsg on errqueue with origin tstamp · 829ae9d6
      Willem de Bruijn 提交于
      Allow reading of timestamps and cmsg at the same time on all relevant
      socket families. One use is to correlate timestamps with egress
      device, by asking for cmsg IP_PKTINFO.
      
      on AF_INET sockets, call the relevant function (ip_cmsg_recv). To
      avoid changing legacy expectations, only do so if the caller sets a
      new timestamping flag SOF_TIMESTAMPING_OPT_CMSG.
      
      on AF_INET6 sockets, IPV6_PKTINFO and all other recv cmsg are already
      returned for all origins. only change is to set ifindex, which is
      not initialized for all error origins.
      
      In both cases, only generate the pktinfo message if an ifindex is
      known. This is not the case for ACK timestamps.
      
      The difference between the protocol families is probably a historical
      accident as a result of the different conditions for generating cmsg
      in the relevant ip(v6)_recv_error function:
      
      ipv4:        if (serr->ee.ee_origin == SO_EE_ORIGIN_ICMP) {
      ipv6:        if (serr->ee.ee_origin != SO_EE_ORIGIN_LOCAL) {
      
      At one time, this was the same test bar for the ICMP/ICMP6
      distinction. This is no longer true.
      Signed-off-by: NWillem de Bruijn <willemb@google.com>
      
      ----
      
      Changes
        v1 -> v2
          large rewrite
          - integrate with existing pktinfo cmsg generation code
          - on ipv4: only send with new flag, to maintain legacy behavior
          - on ipv6: send at most a single pktinfo cmsg
          - on ipv6: initialize fields if not yet initialized
      
      The recv cmsg interfaces are also relevant to the discussion of
      whether looping packet headers is problematic. For v6, cmsgs that
      identify many headers are already returned. This patch expands
      that to v4. If it sounds reasonable, I will follow with patches
      
      1. request timestamps without payload with SOF_TIMESTAMPING_OPT_TSONLY
         (http://patchwork.ozlabs.org/patch/366967/)
      2. sysctl to conditionally drop all timestamps that have payload or
         cmsg from users without CAP_NET_RAW.
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      829ae9d6
  9. 03 12月, 2014 1 次提交
  10. 26 11月, 2014 1 次提交
  11. 25 11月, 2014 1 次提交
    • M
      ipvlan: Initial check-in of the IPVLAN driver. · 2ad7bf36
      Mahesh Bandewar 提交于
      This driver is very similar to the macvlan driver except that it
      uses L3 on the frame to determine the logical interface while
      functioning as packet dispatcher. It inherits L2 of the master
      device hence the packets on wire will have the same L2 for all
      the packets originating from all virtual devices off of the same
      master device.
      
      This driver was developed keeping the namespace use-case in
      mind. Hence most of the examples given here take that as the
      base setup where main-device belongs to the default-ns and
      virtual devices are assigned to the additional namespaces.
      
      The device operates in two different modes and the difference
      in these two modes in primarily in the TX side.
      
      (a) L2 mode : In this mode, the device behaves as a L2 device.
      TX processing upto L2 happens on the stack of the virtual device
      associated with (namespace). Packets are switched after that
      into the main device (default-ns) and queued for xmit.
      
      RX processing is simple and all multicast, broadcast (if
      applicable), and unicast belonging to the address(es) are
      delivered to the virtual devices.
      
      (b) L3 mode : In this mode, the device behaves like a L3 device.
      TX processing upto L3 happens on the stack of the virtual device
      associated with (namespace). Packets are switched to the
      main-device (default-ns) for the L2 processing. Hence the routing
      table of the default-ns will be used in this mode.
      
      RX processins is somewhat similar to the L2 mode except that in
      this mode only Unicast packets are delivered to the virtual device
      while main-dev will handle all other packets.
      
      The devices can be added using the "ip" command from the iproute2
      package -
      
      	ip link add link <master> <virtual> type ipvlan mode [ l2 | l3 ]
      Signed-off-by: NMahesh Bandewar <maheshb@google.com>
      Cc: Eric Dumazet <edumazet@google.com>
      Cc: Maciej Żenczykowski <maze@google.com>
      Cc: Laurent Chavey <chavey@google.com>
      Cc: Tim Hockin <thockin@google.com>
      Cc: Brandon Philips <brandon.philips@coreos.com>
      Cc: Pavel Emelianov <xemul@parallels.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      2ad7bf36
  12. 20 11月, 2014 1 次提交
  13. 06 11月, 2014 1 次提交
  14. 30 10月, 2014 2 次提交
    • E
      net: ipv6: Add a sysctl to make optimistic addresses useful candidates · 7fd2561e
      Erik Kline 提交于
      Add a sysctl that causes an interface's optimistic addresses
      to be considered equivalent to other non-deprecated addresses
      for source address selection purposes.  Preferred addresses
      will still take precedence over optimistic addresses, subject
      to other ranking in the source address selection algorithm.
      
      This is useful where different interfaces are connected to
      different networks from different ISPs (e.g., a cell network
      and a home wifi network).
      
      The current behaviour complies with RFC 3484/6724, and it
      makes sense if the host has only one interface, or has
      multiple interfaces on the same network (same or cooperating
      administrative domain(s), but not in the multiple distinct
      networks case.
      
      For example, if a mobile device has an IPv6 address on an LTE
      network and then connects to IPv6-enabled wifi, while the wifi
      IPv6 address is undergoing DAD, IPv6 connections will try use
      the wifi default route with the LTE IPv6 address, and will get
      stuck until they time out.
      
      Also, because optimistic nodes can receive frames, issue
      an RTM_NEWADDR as soon as DAD starts (with the IFA_F_OPTIMSTIC
      flag appropriately set).  A second RTM_NEWADDR is sent if DAD
      completes (the address flags have changed), otherwise an
      RTM_DELADDR is sent.
      
      Also: add an entry in ip-sysctl.txt for optimistic_dad.
      Signed-off-by: NErik Kline <ek@google.com>
      Acked-by: NLorenzo Colitti <lorenzo@google.com>
      Acked-by: NHannes Frederic Sowa <hannes@stressinduktion.org>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      7fd2561e
    • 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
  15. 11 10月, 2014 1 次提交
  16. 02 10月, 2014 1 次提交
  17. 29 9月, 2014 1 次提交
    • D
      net: tcp: add DCTCP congestion control algorithm · e3118e83
      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>
      e3118e83
  18. 28 9月, 2014 1 次提交
  19. 27 9月, 2014 2 次提交
    • A
      bpf: verifier (add docs) · 51580e79
      Alexei Starovoitov 提交于
      this patch adds all of eBPF verfier documentation and empty bpf_check()
      
      The end goal for the verifier is to statically check safety of the program.
      
      Verifier will catch:
      - loops
      - out of range jumps
      - unreachable instructions
      - invalid instructions
      - uninitialized register access
      - uninitialized stack access
      - misaligned stack access
      - out of range stack access
      - invalid calling convention
      
      More details in Documentation/networking/filter.txt
      Signed-off-by: NAlexei Starovoitov <ast@plumgrid.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      51580e79
    • A
      bpf: introduce BPF syscall and maps · 99c55f7d
      Alexei Starovoitov 提交于
      BPF syscall is a multiplexor for a range of different operations on eBPF.
      This patch introduces syscall with single command to create a map.
      Next patch adds commands to access maps.
      
      'maps' is a generic storage of different types for sharing data between kernel
      and userspace.
      
      Userspace example:
      /* this syscall wrapper creates a map with given type and attributes
       * and returns map_fd on success.
       * use close(map_fd) to delete the map
       */
      int bpf_create_map(enum bpf_map_type map_type, int key_size,
                         int value_size, int max_entries)
      {
          union bpf_attr attr = {
              .map_type = map_type,
              .key_size = key_size,
              .value_size = value_size,
              .max_entries = max_entries
          };
      
          return bpf(BPF_MAP_CREATE, &attr, sizeof(attr));
      }
      
      'union bpf_attr' is backwards compatible with future extensions.
      
      More details in Documentation/networking/filter.txt and in manpage
      Signed-off-by: NAlexei Starovoitov <ast@plumgrid.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      99c55f7d
  20. 26 9月, 2014 4 次提交
  21. 24 9月, 2014 1 次提交
    • E
      icmp: add a global rate limitation · 4cdf507d
      Eric Dumazet 提交于
      Current ICMP rate limiting uses inetpeer cache, which is an RBL tree
      protected by a lock, meaning that hosts can be stuck hard if all cpus
      want to check ICMP limits.
      
      When say a DNS or NTP server process is restarted, inetpeer tree grows
      quick and machine comes to its knees.
      
      iptables can not help because the bottleneck happens before ICMP
      messages are even cooked and sent.
      
      This patch adds a new global limitation, using a token bucket filter,
      controlled by two new sysctl :
      
      icmp_msgs_per_sec - INTEGER
          Limit maximal number of ICMP packets sent per second from this host.
          Only messages whose type matches icmp_ratemask are
          controlled by this limit.
          Default: 1000
      
      icmp_msgs_burst - INTEGER
          icmp_msgs_per_sec controls number of ICMP packets sent per second,
          while icmp_msgs_burst controls the burst size of these packets.
          Default: 50
      
      Note that if we really want to send millions of ICMP messages per
      second, we might extend idea and infra added in commit 04ca6973
      ("ip: make IP identifiers less predictable") :
      add a token bucket in the ip_idents hash and no longer rely on inetpeer.
      Signed-off-by: NEric Dumazet <edumazet@google.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      4cdf507d
  22. 13 9月, 2014 1 次提交
  23. 11 9月, 2014 1 次提交
  24. 10 9月, 2014 1 次提交
    • A
      net: filter: add "load 64-bit immediate" eBPF instruction · 02ab695b
      Alexei Starovoitov 提交于
      add BPF_LD_IMM64 instruction to load 64-bit immediate value into a register.
      All previous instructions were 8-byte. This is first 16-byte instruction.
      Two consecutive 'struct bpf_insn' blocks are interpreted as single instruction:
      insn[0].code = BPF_LD | BPF_DW | BPF_IMM
      insn[0].dst_reg = destination register
      insn[0].imm = lower 32-bit
      insn[1].code = 0
      insn[1].imm = upper 32-bit
      All unused fields must be zero.
      
      Classic BPF has similar instruction: BPF_LD | BPF_W | BPF_IMM
      which loads 32-bit immediate value into a register.
      
      x64 JITs it as single 'movabsq %rax, imm64'
      arm64 may JIT as sequence of four 'movk x0, #imm16, lsl #shift' insn
      
      Note that old eBPF programs are binary compatible with new interpreter.
      
      It helps eBPF programs load 64-bit constant into a register with one
      instruction instead of using two registers and 4 instructions:
      BPF_MOV32_IMM(R1, imm32)
      BPF_ALU64_IMM(BPF_LSH, R1, 32)
      BPF_MOV32_IMM(R2, imm32)
      BPF_ALU64_REG(BPF_OR, R1, R2)
      
      User space generated programs will use this instruction to load constants only.
      
      To tell kernel that user space needs a pointer the _pseudo_ variant of
      this instruction may be added later, which will use extra bits of encoding
      to indicate what type of pointer user space is asking kernel to provide.
      For example 'off' or 'src_reg' fields can be used for such purpose.
      src_reg = 1 could mean that user space is asking kernel to validate and
      load in-kernel map pointer.
      src_reg = 2 could mean that user space needs readonly data section pointer
      src_reg = 3 could mean that user space needs a pointer to per-cpu local data
      All such future pseudo instructions will not be carrying the actual pointer
      as part of the instruction, but rather will be treated as a request to kernel
      to provide one. The kernel will verify the request_for_a_pointer, then
      will drop _pseudo_ marking and will store actual internal pointer inside
      the instruction, so the end result is the interpreter and JITs never
      see pseudo BPF_LD_IMM64 insns and only operate on generic BPF_LD_IMM64 that
      loads 64-bit immediate into a register. User space never operates on direct
      pointers and verifier can easily recognize request_for_pointer vs other
      instructions.
      Signed-off-by: NAlexei Starovoitov <ast@plumgrid.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      02ab695b
  25. 08 9月, 2014 1 次提交
  26. 06 9月, 2014 1 次提交
  27. 05 9月, 2014 2 次提交
  28. 02 9月, 2014 1 次提交
  29. 26 8月, 2014 1 次提交
  30. 03 8月, 2014 3 次提交
    • V
      i40e: adds FCoE to build and updates its documentation · 38758f55
      Vasu Dev 提交于
      Adds newly added FCoE files to the build but only if FCoE module is configured.
      
      Also, updates i40e document for added FCoE support.
      Signed-off-by: NVasu Dev <vasu.dev@intel.com>
      Tested-by: Jack Morgan<jack.morgan@intel.com>
      Signed-off-by: NAaron Brown <aaron.f.brown@intel.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      38758f55
    • A
      net: filter: split 'struct sk_filter' into socket and bpf parts · 7ae457c1
      Alexei Starovoitov 提交于
      clean up names related to socket filtering and bpf in the following way:
      - everything that deals with sockets keeps 'sk_*' prefix
      - everything that is pure BPF is changed to 'bpf_*' prefix
      
      split 'struct sk_filter' into
      struct sk_filter {
      	atomic_t        refcnt;
      	struct rcu_head rcu;
      	struct bpf_prog *prog;
      };
      and
      struct bpf_prog {
              u32                     jited:1,
                                      len:31;
              struct sock_fprog_kern  *orig_prog;
              unsigned int            (*bpf_func)(const struct sk_buff *skb,
                                                  const struct bpf_insn *filter);
              union {
                      struct sock_filter      insns[0];
                      struct bpf_insn         insnsi[0];
                      struct work_struct      work;
              };
      };
      so that 'struct bpf_prog' can be used independent of sockets and cleans up
      'unattached' bpf use cases
      
      split SK_RUN_FILTER macro into:
          SK_RUN_FILTER to be used with 'struct sk_filter *' and
          BPF_PROG_RUN to be used with 'struct bpf_prog *'
      
      __sk_filter_release(struct sk_filter *) gains
      __bpf_prog_release(struct bpf_prog *) helper function
      
      also perform related renames for the functions that work
      with 'struct bpf_prog *', since they're on the same lines:
      
      sk_filter_size -> bpf_prog_size
      sk_filter_select_runtime -> bpf_prog_select_runtime
      sk_filter_free -> bpf_prog_free
      sk_unattached_filter_create -> bpf_prog_create
      sk_unattached_filter_destroy -> bpf_prog_destroy
      sk_store_orig_filter -> bpf_prog_store_orig_filter
      sk_release_orig_filter -> bpf_release_orig_filter
      __sk_migrate_filter -> bpf_migrate_filter
      __sk_prepare_filter -> bpf_prepare_filter
      
      API for attaching classic BPF to a socket stays the same:
      sk_attach_filter(prog, struct sock *)/sk_detach_filter(struct sock *)
      and SK_RUN_FILTER(struct sk_filter *, ctx) to execute a program
      which is used by sockets, tun, af_packet
      
      API for 'unattached' BPF programs becomes:
      bpf_prog_create(struct bpf_prog **)/bpf_prog_destroy(struct bpf_prog *)
      and BPF_PROG_RUN(struct bpf_prog *, ctx) to execute a program
      which is used by isdn, ppp, team, seccomp, ptp, xt_bpf, cls_bpf, test_bpf
      Signed-off-by: NAlexei Starovoitov <ast@plumgrid.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      7ae457c1
    • A
      net: filter: rename sk_chk_filter() -> bpf_check_classic() · 4df95ff4
      Alexei Starovoitov 提交于
      trivial rename to indicate that this functions performs classic BPF checking
      Signed-off-by: NAlexei Starovoitov <ast@plumgrid.com>
      Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      4df95ff4
  31. 31 7月, 2014 1 次提交