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  1. 09 3月, 2015 1 次提交
  3. 07 3月, 2015 1 次提交
  5. 04 3月, 2015 1 次提交
  7. 25 2月, 2015 1 次提交
  9. 24 2月, 2015 4 次提交
  11. 08 2月, 2015 1 次提交
    • N
      tcp: helpers to mitigate ACK loops by rate-limiting out-of-window dupacks · 032ee423
      由 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
  13. 05 2月, 2015 1 次提交
  15. 03 2月, 2015 3 次提交
  17. 27 1月, 2015 1 次提交
    • J
      openvswitch: Add support for unique flow IDs. · 74ed7ab9
      由 Joe Stringer 提交于 
      Previously, flows were manipulated by userspace specifying a full,
unmasked flow key. This adds significant burden onto flow
serialization/deserialization, particularly when dumping flows.

This patch adds an alternative way to refer to flows using a
variable-length "unique flow identifier" (UFID). At flow setup time,
userspace may specify a UFID for a flow, which is stored with the flow
and inserted into a separate table for lookup, in addition to the
standard flow table. Flows created using a UFID must be fetched or
deleted using the UFID.

All flow dump operations may now be made more terse with OVS_UFID_F_*
flags. For example, the OVS_UFID_F_OMIT_KEY flag allows responses to
omit the flow key from a datapath operation if the flow has a
corresponding UFID. This significantly reduces the time spent assembling
and transacting netlink messages. With all OVS_UFID_F_OMIT_* flags
enabled, the datapath only returns the UFID and statistics for each flow
during flow dump, increasing ovs-vswitchd revalidator performance by 40%
or more.
Signed-off-by: NJoe Stringer <joestringer@nicira.com>
Acked-by: NPravin B Shelar <pshelar@nicira.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>
      74ed7ab9
  19. 26 1月, 2015 1 次提交
  21. 14 1月, 2015 1 次提交
  23. 13 1月, 2015 1 次提交
  25. 12 1月, 2015 1 次提交
    • W
      doc: fix the compile fix of txtimestamp.c · d3b4b261
      由 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
  27. 09 1月, 2015 1 次提交
  29. 30 12月, 2014 1 次提交
  31. 23 12月, 2014 1 次提交
  33. 16 12月, 2014 1 次提交
  35. 10 12月, 2014 1 次提交
  37. 09 12月, 2014 2 次提交
    • W
      net-timestamp: expand documentation and test · cbd3aad5
      由 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
      由 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
  39. 03 12月, 2014 1 次提交
  41. 26 11月, 2014 1 次提交
  43. 25 11月, 2014 1 次提交
    • M
      ipvlan: Initial check-in of the IPVLAN driver. · 2ad7bf36
      由 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
  45. 20 11月, 2014 1 次提交
  47. 06 11月, 2014 1 次提交
  49. 30 10月, 2014 2 次提交
    • E
      net: ipv6: Add a sysctl to make optimistic addresses useful candidates · 7fd2561e
      由 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
      由 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
  51. 11 10月, 2014 1 次提交
  53. 02 10月, 2014 1 次提交
  55. 29 9月, 2014 1 次提交
    • D
      net: tcp: add DCTCP congestion control algorithm · e3118e83
      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>
      e3118e83
  57. 28 9月, 2014 1 次提交
  59. 27 9月, 2014 2 次提交
    • A
      bpf: verifier (add docs) · 51580e79
      由 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
      由 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
  61. 26 9月, 2014 2 次提交