path: root/net/ipv4/tcp_rate.c (follow)
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2019-05-21treewide: Add SPDX license identifier for missed filesThomas Gleixner1-0/+1
Add SPDX license identifiers to all files which: - Have no license information of any form - Have EXPORT_.*_SYMBOL_GPL inside which was used in the initial scan/conversion to ignore the file These files fall under the project license, GPL v2 only. The resulting SPDX license identifier is: GPL-2.0-only Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2018-09-21tcp: introduce tcp_skb_timestamp_us() helperEric Dumazet1-7/+8
There are few places where TCP reads skb->skb_mstamp expecting a value in usec unit. skb->tstamp (aka skb->skb_mstamp) will soon store CLOCK_TAI nsec value. Add tcp_skb_timestamp_us() to provide proper conversion when needed. Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2018-07-11tcp: expose both send and receive intervals for rate sampleDeepti Raghavan1-0/+4
Congestion control algorithms, which access the rate sample through the tcp_cong_control function, only have access to the maximum of the send and receive interval, for cases where the acknowledgment rate may be inaccurate due to ACK compression or decimation. Algorithms may want to use send rates and receive rates as separate signals. Signed-off-by: Deepti Raghavan <deeptir@mit.edu> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-12-08tcp: invalidate rate samples during SACK renegingYousuk Seung1-3/+7
Mark tcp_sock during a SACK reneging event and invalidate rate samples while marked. Such rate samples may overestimate bw by including packets that were SACKed before reneging. < ack 6001 win 10000 sack 7001:38001 < ack 7001 win 0 sack 8001:38001 // Reneg detected > seq 7001:8001 // RTO, SACK cleared. < ack 38001 win 10000 In above example the rate sample taken after the last ack will count 7001-38001 as delivered while the actual delivery rate likely could be much lower i.e. 7001-8001. This patch adds a new field tcp_sock.sack_reneg and marks it when we declare SACK reneging and entering TCP_CA_Loss, and unmarks it after the last rate sample was taken before moving back to TCP_CA_Open. This patch also invalidates rate samples taken while tcp_sock.is_sack_reneg is set. Fixes: b9f64820fb22 ("tcp: track data delivery rate for a TCP connection") Signed-off-by: Yousuk Seung <ysseung@google.com> Signed-off-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Yuchung Cheng <ycheng@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Acked-by: Eric Dumazet <edumazet@google.com> Acked-by: Priyaranjan Jha <priyarjha@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-06-15tcp: export do_tcp_sendpages and tcp_rate_check_app_limited functionsDave Watson1-0/+1
Export do_tcp_sendpages and tcp_rate_check_app_limited, since tls will need to sendpages while the socket is already locked. tcp_sendpage is exported, but requires the socket lock to not be held already. Signed-off-by: Aviad Yehezkel <aviadye@mellanox.com> Signed-off-by: Ilya Lesokhin <ilyal@mellanox.com> Signed-off-by: Boris Pismenny <borisp@mellanox.com> Signed-off-by: Dave Watson <davejwatson@fb.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-05-17tcp: switch TCP TS option (RFC 7323) to 1ms clockEric Dumazet1-8/+8
TCP Timestamps option is defined in RFC 7323 Traditionally on linux, it has been tied to the internal 'jiffies' variable, because it had been a cheap and good enough generator. For TCP flows on the Internet, 1 ms resolution would be much better than 4ms or 10ms (HZ=250 or HZ=100 respectively) For TCP flows in the DC, Google has used usec resolution for more than two years with great success [1] Receive size autotuning (DRS) is indeed more precise and converges faster to optimal window size. This patch converts tp->tcp_mstamp to a plain u64 value storing a 1 usec TCP clock. This choice will allow us to upstream the 1 usec TS option as discussed in IETF 97. [1] https://www.ietf.org/proceedings/97/slides/slides-97-tcpm-tcp-options-for-low-latency-00.pdf Signed-off-by: Eric Dumazet <edumazet@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2017-04-26tcp: do not pass timestamp to tcp_rate_gen()Eric Dumazet1-3/+4
No longer needed, since tp->tcp_mstamp holds the information. This is needed to remove sack_state.ack_time in a following patch. Signed-off-by: Eric Dumazet <edumazet@google.com> Acked-by: Soheil Hassas Yeganeh <soheil@google.com> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
2016-09-21tcp: export data delivery rateYuchung Cheng1-1/+11
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: 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: David S. Miller <davem@davemloft.net>
2016-09-21tcp: track application-limited rate samplesSoheil Hassas Yeganeh1-1/+28
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: 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: David S. Miller <davem@davemloft.net>
2016-09-21tcp: track data delivery rate for a TCP connectionYuchung Cheng1-0/+149
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: 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: David S. Miller <davem@davemloft.net>