- 22 Sep, 2016 6 commits
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David Howells authored
Add a ktime_sub_ms() to go with ktime_add_ms() and co. for use in AF_RXRPC RTT determination. Signed-off-by:David Howells <dhowells@redhat.com>
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David Howells authored
Expedite the transmission of a response to a PING ACK by sending it from sendmsg if one is pending. We're most likely to see a PING ACK during the client call Tx phase as the other side may use it to determine a number of parameters, such as the client's receive window size, the RTT and whether the client is doing slow start (similar to RFC5681). If we don't expedite it, it's left to the background processing thread to transmit. Signed-off-by: -
David Howells authored
Send a PING ACK packet to the peer when we get a new incoming call from a peer we don't have a record for. The PING RESPONSE ACK packet will tell us the following about the peer: (1) its receive window size (2) its MTU sizes (3) its support for jumbo DATA packets (4) if it supports slow start (similar to RFC 5681) (5) an estimate of the RTT This is necessary because the peer won't normally send us an ACK until it gets to the Rx phase and we send it a packet, but we would like to know some of this information before we start sending packets. A pair of tracepoints are added so that RTT determination can be observed. Signed-off-by: -
David Howells authored
Add a function to track the average RTT for a peer. Sources of RTT data will be added in subsequent patches. The RTT data will be useful in the future for determining resend timeouts and for handling the slow-start part of the Rx protocol. Also add a pair of tracepoints, one to log transmissions to elicit a response for RTT purposes and one to log responses that contribute RTT data. Signed-off-by: -
David Howells authored
Add a Tx-phase annotation for packet buffers to indicate that a buffer has already been retransmitted. This will be used by future congestion management. Re-retransmissions of a packet don't affect the congestion window managment in the same way as initial retransmissions. Signed-off-by: -
David Howells authored
Don't store the rxrpc protocol header in sk_buffs on the transmit queue, but rather generate it on the fly and pass it to kernel_sendmsg() as a separate iov. This reduces the amount of storage required. Note that the security header is still stored in the sk_buff as it may get encrypted along with the data (and doesn't change with each transmission). Signed-off-by:
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- 21 Sep, 2016 34 commits
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David S. Miller authored
Jiri Pirko says: ==================== mlxsw: Replace Hw related const with resource query results Nogah says: Many of the ASIC's properties can be read from the HW with resources query. This patchset adds new resources to the resource query and implement using them, instead of the constants that we currently use. Those resources are lag, kvd and router related. ==================== Signed-off-by:David S. Miller <davem@davemloft.net>
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Nogah Frankel authored
Replace max rif const with using the result from resource query. Signed-off-by:
Nogah Frankel <nogahf@mellanox.com> Reviewed-by:
Ido Schimmel <idosch@mellanox.com> Signed-off-by:
Jiri Pirko <jiri@mellanox.com> Signed-off-by:
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Nogah Frankel authored
Add the max number of rif (router interfaces) to resource query. Signed-off-by:
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Nogah Frankel authored
Add max system ports, max regions and max vlan groups to resource query. Signed-off-by:
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Nogah Frankel authored
Replace max virtual routers const with the result from the resource query. Signed-off-by:
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Nogah Frankel authored
Add the max number of virtual routers to resource query. Signed-off-by:
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Nogah Frankel authored
Use resources from resource query to determine values for the profile configuration. Add KVD determined section sizes to the resources struct. Change the profile struct and value to match this changes. Signed-off-by:
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Nogah Frankel authored
Add KVD size, and minimum sizes for the single and double sections resources to resources query. Signed-off-by:
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Nogah Frankel authored
Use max lag and max ports in lag resources as the result of resource query instead of using const to save them. Signed-off-by:
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Nogah Frankel authored
Add max lag and max ports in lag resources to resources query. Signed-off-by:
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Or Gerlitz authored
The offloads stats functions are local to this file, make them static. Fixes: fc1bbb0f ('mlxsw: spectrum: Implement offload stats ndo [..]') Signed-off-by:
Or Gerlitz <ogerlitz@mellanox.com> Acked-by:
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David S. Miller authored
Neal Cardwell says: ==================== tcp: BBR congestion control algorithm This patch series implements a new TCP congestion control algorithm: BBR (Bottleneck Bandwidth and RTT). A paper with a detailed description of BBR will be published in ACM Queue, September-October 2016, as "BBR: Congestion-Based Congestion Control". BBR is widely deployed in production at Google. The patch series starts with a set of supporting infrastructure changes, including a few that extend the congestion control framework. The last patch adds BBR as a TCP congestion control module. Please see individual patches for the details. - v3 -> v4: - Updated tcp_bbr.c in "tcp_bbr: add BBR congestion control" to use const to qualify all the constant parameters. Thanks to Stephen Hemminger. - In "tcp_bbr: add BBR congestion control", remove the bbr_rate_kbps() function, which had a 64-bit divide that would be problematic on some architectures, and just use bbr_rate_bytes_per_sec() directly. Thanks to Kenneth Klette Jonassen for suggesting this. - In "tcp: switch back to proper tcp_skb_cb size check in tcp_init()", switched from sizeof(skb->cb) to FIELD_SIZEOF. Thanks to Lance Richardson for suggesting this. - Updated "tcp_bbr: add BBR congestion control" commit message with performance data, more details about deployment at Google, and another reminder to use fq with BBR. - Updated tcp_bbr.c in "tcp_bbr: add BBR congestion control" to use MODULE_LICENSE("Dual BSD/GPL"). - v2 -> v3: fix another issue caught by build bots: - adjust rate_sample struct initialization syntax to allow gcc-4.4 to compile the "tcp: track data delivery rate for a TCP connection" patch; also adjusted some similar syntax in "tcp_bbr: add BBR congestion control" - v1 -> v2: fix issues caught by build bots: - fix "tcp: export data delivery rate" to use rate64 instead of rate, so there is a 64-bit numerator for the do_div call - fix conflicting definitions for minmax caused by "tcp: use windowed min filter library for TCP min_rtt estimation" with a new commit: tcp: cdg: rename struct minmax in tcp_cdg.c to avoid a naming conflict - fix warning about the use of __packed in "tcp: track data delivery rate for a TCP connection", which involves the addition of a new commit: tcp: switch back to proper tcp_skb_cb size check in tcp_init() ==================== Signed-off-by: -
Neal Cardwell authored
This commit implements a new TCP congestion control algorithm: BBR (Bottleneck Bandwidth and RTT). A detailed description of BBR will be published in ACM Queue, Vol. 14 No. 5, September-October 2016, as "BBR: Congestion-Based Congestion Control". BBR has significantly increased throughput and reduced latency for connections on Google's internal backbone networks and google.com and YouTube Web servers. BBR requires only changes on the sender side, not in the network or the receiver side. Thus it can be incrementally deployed on today's Internet, or in datacenters. The Internet has predominantly used loss-based congestion control (largely Reno or CUBIC) since the 1980s, relying on packet loss as the signal to slow down. While this worked well for many years, loss-based congestion control is unfortunately out-dated in today's networks. On today's Internet, loss-based congestion control causes the infamous bufferbloat problem, often causing seconds of needless queuing delay, since it fills the bloated buffers in many last-mile links. On today's high-speed long-haul links using commodity switches with shallow buffers, loss-based congestion control has abysmal throughput because it over-reacts to losses caused by transient traffic bursts. In 1981 Kleinrock and Gale showed that the optimal operating point for a network maximizes delivered bandwidth while minimizing delay and loss, not only for single connections but for the network as a whole. Finding that optimal operating point has been elusive, since any single network measurement is ambiguous: network measurements are the result of both bandwidth and propagation delay, and those two cannot be measured simultaneously. While it is impossible to disambiguate any single bandwidth or RTT measurement, a connection's behavior over time tells a clearer story. BBR uses a measurement strategy designed to resolve this ambiguity. It combines these measurements with a robust servo loop using recent control systems advances to implement a distributed congestion control algorithm that reacts to actual congestion, not packet loss or transient queue delay, and is designed to converge with high probability to a point near the optimal operating point. In a nutshell, BBR creates an explicit model of the network pipe by sequentially probing the bottleneck bandwidth and RTT. On the arrival of each ACK, BBR derives the current delivery rate of the last round trip, and feeds it through a windowed max-filter to estimate the bottleneck bandwidth. Conversely it uses a windowed min-filter to estimate the round trip propagation delay. The max-filtered bandwidth and min-filtered RTT estimates form BBR's model of the network pipe. Using its model, BBR sets control parameters to govern sending behavior. The primary control is the pacing rate: BBR applies a gain multiplier to transmit faster or slower than the observed bottleneck bandwidth. The conventional congestion window (cwnd) is now the secondary control; the cwnd is set to a small multiple of the estimated BDP (bandwidth-delay product) in order to allow full utilization and bandwidth probing while bounding the potential amount of queue at the bottleneck. When a BBR connection starts, it enters STARTUP mode and applies a high gain to perform an exponential search to quickly probe the bottleneck bandwidth (doubling its sending rate each round trip, like slow start). However, instead of continuing until it fills up the buffer (i.e. a loss), or until delay or ACK spacing reaches some threshold (like Hystart), it uses its model of the pipe to estimate when that pipe is full: it estimates the pipe is full when it notices the estimated bandwidth has stopped growing. At that point it exits STARTUP and enters DRAIN mode, where it reduces its pacing rate to drain the queue it estimates it has created. Then BBR enters steady state. In steady state, PROBE_BW mode cycles between first pacing faster to probe for more bandwidth, then pacing slower to drain any queue that created if no more bandwidth was available, and then cruising at the estimated bandwidth to utilize the pipe without creating excess queue. Occasionally, on an as-needed basis, it sends significantly slower to probe for RTT (PROBE_RTT mode). BBR has been fully deployed on Google's wide-area backbone networks and we're experimenting with BBR on Google.com and YouTube on a global scale. Replacing CUBIC with BBR has resulted in significant improvements in network latency and application (RPC, browser, and video) metrics. For more details please refer to our upcoming ACM Queue publication. Example performance results, to illustrate the difference between BBR and CUBIC: Resilience to random loss (e.g. from shallow buffers): Consider a netperf TCP_STREAM test lasting 30 secs on an emulated path with a 10Gbps bottleneck, 100ms RTT, and 1% packet loss rate. CUBIC gets 3.27 Mbps, and BBR gets 9150 Mbps (2798x higher). Low latency with the bloated buffers common in today's last-mile links: Consider a netperf TCP_STREAM test lasting 120 secs on an emulated path with a 10Mbps bottleneck, 40ms RTT, and 1000-packet bottleneck buffer. Both fully utilize the bottleneck bandwidth, but BBR achieves this with a median RTT 25x lower (43 ms instead of 1.09 secs). Our long-term goal is to improve the congestion control algorithms used on the Internet. We are hopeful that BBR can help advance the efforts toward this goal, and motivate the community to do further research. Test results, performance evaluations, feedback, and BBR-related discussions are very welcome in the public e-mail list for BBR: https://groups.google.com/forum/#!forum/bbr-dev NOTE: BBR *must* be used with the fq qdisc ("man tc-fq") with pacing enabled, since pacing is integral to the BBR design and implementation. BBR without pacing would not function properly, and may incur unnecessary high packet loss rates. Signed-off-by:
Van Jacobson <vanj@google.com> Signed-off-by:
Neal Cardwell <ncardwell@google.com> Signed-off-by:
Yuchung Cheng <ycheng@google.com> Signed-off-by:
Nandita Dukkipati <nanditad@google.com> Signed-off-by:
Eric Dumazet <edumazet@google.com> Signed-off-by:
Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by:
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Neal Cardwell authored
The TCP CUBIC module already uses 64 bytes. The upcoming TCP BBR module uses 88 bytes. Signed-off-by:
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Yuchung Cheng authored
This commit introduces an optional new "omnipotent" hook, cong_control(), for congestion control modules. The cong_control() function is called at the end of processing an ACK (i.e., after updating sequence numbers, the SACK scoreboard, and loss detection). At that moment we have precise delivery rate information the congestion control module can use to control the sending behavior (using cwnd, TSO skb size, and pacing rate) in any CA state. This function can also be used by a congestion control that prefers not to use the default cwnd reduction approach (i.e., the PRR algorithm) during CA_Recovery to control the cwnd and sending rate during loss recovery. We take advantage of the fact that recent changes defer the retransmission or transmission of new data (e.g. by F-RTO) in recovery until the new tcp_cong_control() function is run. With this commit, we only run tcp_update_pacing_rate() if the congestion control is not using this new API. New congestion controls which use the new API do not want the TCP stack to run the default pacing rate calculation and overwrite whatever pacing rate they have chosen at initialization time. Signed-off-by:
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Yuchung Cheng authored
Currently the TCP send buffer expands to twice cwnd, in order to allow limited transmits in the CA_Recovery state. This assumes that cwnd does not increase in the CA_Recovery. For some congestion control algorithms, like the upcoming BBR module, if the losses in recovery do not indicate congestion then we may continue to raise cwnd multiplicatively in recovery. In such cases the current multiplier will falsely limit the sending rate, much as if it were limited by the application. This commit adds an optional congestion control callback to use a different multiplier to expand the TCP send buffer. For congestion control modules that do not specificy this callback, TCP continues to use the previous default of 2. Signed-off-by:
Stephen Hemminger <stephen@networkplumber.org> Signed-off-by:
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Neal Cardwell authored
Export tcp_mss_to_mtu(), so that congestion control modules can use this to help calculate a pacing rate. Signed-off-by:
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Neal Cardwell authored
To allow congestion control modules to use the default TSO auto-sizing algorithm as one of the ingredients in their own decision about TSO sizing: 1) Export tcp_tso_autosize() so that CC modules can use it. 2) Change tcp_tso_autosize() to allow callers to specify a minimum number of segments per TSO skb, in case the congestion control module has a different notion of the best floor for TSO skbs for the connection right now. For very low-rate paths or policed connections it can be appropriate to use smaller TSO skbs. Signed-off-by:
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Neal Cardwell authored
Add the tso_segs_goal() function in tcp_congestion_ops to allow the congestion control module to specify the number of segments that should be in a TSO skb sent by tcp_write_xmit() and tcp_xmit_retransmit_queue(). The congestion control module can either request a particular number of segments in TSO skb that we transmit, or return 0 if it doesn't care. This allows the upcoming BBR congestion control module to select small TSO skb sizes if the module detects that the bottleneck bandwidth is very low, or that the connection is policed to a low rate. Signed-off-by:
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Yuchung Cheng authored
This commit export two new fields in struct tcp_info: tcpi_delivery_rate: The most recent goodput, as measured by tcp_rate_gen(). If the socket is limited by the sending application (e.g., no data to send), it reports the highest measurement instead of the most recent. The unit is bytes per second (like other rate fields in tcp_info). tcpi_delivery_rate_app_limited: A boolean indicating if the goodput was measured when the socket's throughput was limited by the sending application. This delivery rate information can be useful for applications that want to know the current throughput the TCP connection is seeing, e.g. adaptive bitrate video streaming. It can also be very useful for debugging or troubleshooting. Signed-off-by: -
Soheil Hassas Yeganeh authored
This commit adds code to track whether the delivery rate represented by each rate_sample was limited by the application. Upon each transmit, we store in the is_app_limited field in the skb a boolean bit indicating whether there is a known "bubble in the pipe": a point in the rate sample interval where the sender was application-limited, and did not transmit even though the cwnd and pacing rate allowed it. This logic marks the flow app-limited on a write if *all* of the following are true: 1) There is less than 1 MSS of unsent data in the write queue available to transmit. 2) There is no packet in the sender's queues (e.g. in fq or the NIC tx queue). 3) The connection is not limited by cwnd. 4) There are no lost packets to retransmit. The tcp_rate_check_app_limited() code in tcp_rate.c determines whether the connection is application-limited at the moment. If the flow is application-limited, it sets the tp->app_limited field. If the flow is application-limited then that means there is effectively a "bubble" of silence in the pipe now, and this silence will be reflected in a lower bandwidth sample for any rate samples from now until we get an ACK indicating this bubble has exited the pipe: specifically, until we get an ACK for the next packet we transmit. When we send every skb we record in scb->tx.is_app_limited whether the resulting rate sample will be application-limited. The code in tcp_rate_gen() checks to see when it is safe to mark all known application-limited bubbles of silence as having exited the pipe. It does this by checking to see when the delivered count moves past the tp->app_limited marker. At this point it zeroes the tp->app_limited marker, as all known bubbles are out of the pipe. We make room for the tx.is_app_limited bit in the skb by borrowing a bit from the in_flight field used by NV to record the number of bytes in flight. The receive window in the TCP header is 16 bits, and the max receive window scaling shift factor is 14 (RFC 1323). So the max receive window offered by the TCP protocol is 2^(16+14) = 2^30. So we only need 30 bits for the tx.in_flight used by NV. Signed-off-by: -
Yuchung Cheng authored
This patch generates data delivery rate (throughput) samples on a per-ACK basis. These rate samples can be used by congestion control modules, and specifically will be used by TCP BBR in later patches in this series. Key state: tp->delivered: Tracks the total number of data packets (original or not) delivered so far. This is an already-existing field. tp->delivered_mstamp: the last time tp->delivered was updated. Algorithm: A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis: d1: the current tp->delivered after processing the ACK t1: the current time after processing the ACK d0: the prior tp->delivered when the acked skb was transmitted t0: the prior tp->delivered_mstamp when the acked skb was transmitted When an skb is transmitted, we snapshot d0 and t0 in its control block in tcp_rate_skb_sent(). When an ACK arrives, it may SACK and ACK some skbs. For each SACKed or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct to reflect the latest (d0, t0). Finally, tcp_rate_gen() generates a rate sample by storing (d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us. One caveat: if an skb was sent with no packets in flight, then tp->delivered_mstamp may be either invalid (if the connection is starting) or outdated (if the connection was idle). In that case, we'll re-stamp tp->delivered_mstamp. At first glance it seems t0 should always be the time when an skb was transmitted, but actually this could over-estimate the rate due to phase mismatch between transmit and ACK events. To track the delivery rate, we ensure that if packets are in flight then t0 and and t1 are times at which packets were marked delivered. If the initial and final RTTs are different then one may be corrupted by some sort of noise. The noise we see most often is sending gaps caused by delayed, compressed, or stretched acks. This either affects both RTTs equally or artificially reduces the final RTT. We approach this by recording the info we need to compute the initial RTT (duration of the "send phase" of the window) when we recorded the associated inflight. Then, for a filter to avoid bandwidth overestimates, we generalize the per-sample bandwidth computation from: bw = delivered / ack_phase_rtt to the following: bw = delivered / max(send_phase_rtt, ack_phase_rtt) In large-scale experiments, this filtering approach incorporating send_phase_rtt is effective at avoiding bandwidth overestimates due to ACK compression or stretched ACKs. Signed-off-by: -
Neal Cardwell authored
Count the number of packets that a TCP connection marks lost. Congestion control modules can use this loss rate information for more intelligent decisions about how fast to send. Specifically, this is used in TCP BBR policer detection. BBR uses a high packet loss rate as one signal in its policer detection and policer bandwidth estimation algorithm. The BBR policer detection algorithm cannot simply track retransmits, because a retransmit can be (and often is) an indicator of packets lost long, long ago. This is particularly true in a long CA_Loss period that repairs the initial massive losses when a policer kicks in. Signed-off-by:
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Eric Dumazet authored
Revert to the tcp_skb_cb size check that tcp_init() had before commit b4772ef8 ("net: use common macro for assering skb->cb[] available size in protocol families"). As related commit 744d5a3e ("net: move skb->dropcount to skb->cb[]") explains, the sock_skb_cb_check_size() mechanism was added to ensure that there is space for dropcount, "for protocol families using it". But TCP is not a protocol using dropcount, so tcp_init() doesn't need to provision space for dropcount in the skb->cb[], and thus we can revert to the older form of the tcp_skb_cb size check. Doing so allows TCP to use 4 more bytes of the skb->cb[] space. Fixes: b4772ef8 ("net: use common macro for assering skb->cb[] available size in protocol families") Signed-off-by:
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Eric Dumazet authored
This commit adds to the fq module a low_rate_threshold parameter to insert a delay after all packets if the socket requests a pacing rate below the threshold. This helps achieve more precise control of the sending rate with low-rate paths, especially policers. The basic issue is that if a congestion control module detects a policer at a certain rate, it may want fq to be able to shape to that policed rate. That way the sender can avoid policer drops by having the packets arrive at the policer at or just under the policed rate. The default threshold of 550Kbps was chosen analytically so that for policers or links at 500Kbps or 512Kbps fq would very likely invoke this mechanism, even if the pacing rate was briefly slightly above the available bandwidth. This value was then empirically validated with two years of production testing on YouTube video servers. Signed-off-by:
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Neal Cardwell authored
Refactor the TCP min_rtt code to reuse the new win_minmax library in lib/win_minmax.c to simplify the TCP code. This is a pure refactor: the functionality is exactly the same. We just moved the windowed min code to make TCP easier to read and maintain, and to allow other parts of the kernel to use the windowed min/max filter code. Signed-off-by:
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Neal Cardwell authored
This commit introduces a generic library to estimate either the min or max value of a time-varying variable over a recent time window. This is code originally from Kathleen Nichols. The current form of the code is from Van Jacobson. A single struct minmax_sample will track the estimated windowed-max value of the series if you call minmax_running_max() or the estimated windowed-min value of the series if you call minmax_running_min(). Nearly equivalent code is already in place for minimum RTT estimation in the TCP stack. This commit extracts that code and generalizes it to handle both min and max. Moving the code here reduces the footprint and complexity of the TCP code base and makes the filter generally available for other parts of the codebase, including an upcoming TCP congestion control module. This library works well for time series where the measurements are smoothly increasing or decreasing. Signed-off-by:
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Soheil Hassas Yeganeh authored
The upcoming change "lib/win_minmax: windowed min or max estimator" introduces a struct called minmax, which is then included in include/linux/tcp.h in the upcoming change "tcp: use windowed min filter library for TCP min_rtt estimation". This would create a compilation error for tcp_cdg.c, which defines its own minmax struct. To avoid this naming conflict (and potentially others in the future), this commit renames the version used in tcp_cdg.c to cdg_minmax. Signed-off-by:
Kenneth Klette Jonassen <kennetkl@ifi.uio.no> Signed-off-by:
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Sean Wang authored
data_src is unchanged inside the loop, so this patch moves the assignment to outside the loop to avoid unnecessarily assignment Signed-off-by:
Sean Wang <sean.wang@mediatek.com> Signed-off-by:
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Vivien Didelot authored
An address can be loaded in the ATU with multiple ports, for instance when adding multiple ports to a Multicast group with "bridge mdb". The current code doesn't allow that. Add an helper to get a single entry from the ATU, then set or clear the requested port, before loading the entry back in the ATU. Note that the required _mv88e6xxx_atu_getnext function is defined below mv88e6xxx_port_db_load_purge, so forward-declare it for the moment. The ATU code will be isolated in future patches. Fixes: 83dabd1f ("net: dsa: mv88e6xxx: make switchdev DB ops generic") Signed-off-by:
Vivien Didelot <vivien.didelot@savoirfairelinux.com> Reviewed-by:
Andrew Lunn <andrew@lunn.ch> Reviewed-by:
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Jamal Hadi Salim authored
With the batch changes that translated transient actions into a temporary list lost in the translation was the fact that tcf_action_destroy() will eventually delete the action from the permanent location if the refcount is zero. Example of what broke: ...add a gact action to drop sudo $TC actions add action drop index 10 ...now retrieve it, looks good sudo $TC actions get action gact index 10 ...retrieve it again and find it is gone! sudo $TC actions get action gact index 10 Fixes: 22dc13c8 ("net_sched: convert tcf_exts from list to pointer array"), Fixes: 824a7e88 ("net_sched: remove an unnecessary list_del()") Fixes: f07fed82 ("net_sched: remove the leftover cleanup_a()") Acked-by:
Cong Wang <xiyou.wangcong@gmail.com> Signed-off-by:
Jamal Hadi Salim <jhs@mojatatu.com> Signed-off-by:
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David S. Miller authored
Daniel Borkmann says: ==================== BPF direct packet access improvements This set adds write support to the currently available read support for {cls,act}_bpf programs. First one is a fix for affected commit sitting in net-next and prerequisite for the second one, last patch adds a number of test cases against the verifier. For details, please see individual patches. ==================== Signed-off-by: -
Daniel Borkmann authored
Add couple of test cases for direct write and the negative size issue, and also adjust the direct packet access test4 since it asserts that writes are not possible, but since we've just added support for writes, we need to invert the verdict to ACCEPT, of course. Summary: 133 PASSED, 0 FAILED. Signed-off-by:
Daniel Borkmann <daniel@iogearbox.net> Acked-by:
Alexei Starovoitov <ast@kernel.org> Signed-off-by:
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Daniel Borkmann authored
This work implements direct packet access for helpers and direct packet write in a similar fashion as already available for XDP types via commits 4acf6c0b ("bpf: enable direct packet data write for xdp progs") and 6841de8b ("bpf: allow helpers access the packet directly"), and as a complementary feature to the already available direct packet read for tc (cls/act) programs. For enabling this, we need to introduce two helpers, bpf_skb_pull_data() and bpf_csum_update(). The first is generally needed for both, read and write, because they would otherwise only be limited to the current linear skb head. Usually, when the data_end test fails, programs just bail out, or, in the direct read case, use bpf_skb_load_bytes() as an alternative to overcome this limitation. If such data sits in non-linear parts, we can just pull them in once with the new helper, retest and eventually access them. At the same time, this also makes sure the skb is uncloned, which is, of course, a necessary condition for direct write. As this needs to be an invariant for the write part only, the verifier detects writes and adds a prologue that is calling bpf_skb_pull_data() to effectively unclone the skb from the very beginning in case it is indeed cloned. The heuristic makes use of a similar trick that was done in 233577a2 ("net: filter: constify detection of pkt_type_offset"). This comes at zero cost for other programs that do not use the direct write feature. Should a program use this feature only sparsely and has read access for the most parts with, for example, drop return codes, then such write action can be delegated to a tail called program for mitigating this cost of potential uncloning to a late point in time where it would have been paid similarly with the bpf_skb_store_bytes() as well. Advantage of direct write is that the writes are inlined whereas the helper cannot make any length assumptions and thus needs to generate a call to memcpy() also for small sizes, as well as cost of helper call itself with sanity checks are avoided. Plus, when direct read is already used, we don't need to cache or perform rechecks on the data boundaries (due to verifier invalidating previous checks for helpers that change skb->data), so more complex programs using rewrites can benefit from switching to direct read plus write. For direct packet access to helpers, we save the otherwise needed copy into a temp struct sitting on stack memory when use-case allows. Both facilities are enabled via may_access_direct_pkt_data() in verifier. For now, we limit this to map helpers and csum_diff, and can successively enable other helpers where we find it makes sense. Helpers that definitely cannot be allowed for this are those part of bpf_helper_changes_skb_data() since they can change underlying data, and those that write into memory as this could happen for packet typed args when still cloned. bpf_csum_update() helper accommodates for the fact that we need to fixup checksum_complete when using direct write instead of bpf_skb_store_bytes(), meaning the programs can use available helpers like bpf_csum_diff(), and implement csum_add(), csum_sub(), csum_block_add(), csum_block_sub() equivalents in eBPF together with the new helper. A usage example will be provided for iproute2's examples/bpf/ directory. Signed-off-by:
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