tcp: track data delivery rate for a TCP connection
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starting) or outdated (if the connection was idle).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>
diff --git a/net/ipv4/tcp_rate.c b/net/ipv4/tcp_rate.c
new file mode 100644
index 0000000..1daed6a
--- /dev/null
+++ b/net/ipv4/tcp_rate.c
@@ -0,0 +1,149 @@
+#include <net/tcp.h>
+
+/* The bandwidth estimator estimates the rate at which the network
+ * can currently deliver outbound data packets for this flow. At a high
+ * level, it operates by taking a delivery rate sample for each ACK.
+ *
+ * A rate sample records the rate at which the network delivered packets
+ * for this flow, calculated over the time interval between the transmission
+ * of a data packet and the acknowledgment of that packet.
+ *
+ * Specifically, over the interval between each transmit and corresponding ACK,
+ * the estimator generates a delivery rate sample. Typically it uses the rate
+ * at which packets were acknowledged. However, the approach of using only the
+ * acknowledgment rate faces a challenge under the prevalent ACK decimation or
+ * compression: packets can temporarily appear to be delivered much quicker
+ * than the bottleneck rate. Since it is physically impossible to do that in a
+ * sustained fashion, when the estimator notices that the ACK rate is faster
+ * than the transmit rate, it uses the latter:
+ *
+ * send_rate = #pkts_delivered/(last_snd_time - first_snd_time)
+ * ack_rate = #pkts_delivered/(last_ack_time - first_ack_time)
+ * bw = min(send_rate, ack_rate)
+ *
+ * Notice the estimator essentially estimates the goodput, not always the
+ * network bottleneck link rate when the sending or receiving is limited by
+ * other factors like applications or receiver window limits. The estimator
+ * deliberately avoids using the inter-packet spacing approach because that
+ * approach requires a large number of samples and sophisticated filtering.
+ */
+
+
+/* Snapshot the current delivery information in the skb, to generate
+ * a rate sample later when the skb is (s)acked in tcp_rate_skb_delivered().
+ */
+void tcp_rate_skb_sent(struct sock *sk, struct sk_buff *skb)
+{
+ struct tcp_sock *tp = tcp_sk(sk);
+
+ /* In general we need to start delivery rate samples from the
+ * time we received the most recent ACK, to ensure we include
+ * the full time the network needs to deliver all in-flight
+ * packets. If there are no packets in flight yet, then we
+ * know that any ACKs after now indicate that the network was
+ * able to deliver those packets completely in the sampling
+ * interval between now and the next ACK.
+ *
+ * Note that we use packets_out instead of tcp_packets_in_flight(tp)
+ * because the latter is a guess based on RTO and loss-marking
+ * heuristics. We don't want spurious RTOs or loss markings to cause
+ * a spuriously small time interval, causing a spuriously high
+ * bandwidth estimate.
+ */
+ if (!tp->packets_out) {
+ tp->first_tx_mstamp = skb->skb_mstamp;
+ tp->delivered_mstamp = skb->skb_mstamp;
+ }
+
+ TCP_SKB_CB(skb)->tx.first_tx_mstamp = tp->first_tx_mstamp;
+ TCP_SKB_CB(skb)->tx.delivered_mstamp = tp->delivered_mstamp;
+ TCP_SKB_CB(skb)->tx.delivered = tp->delivered;
+}
+
+/* When an skb is sacked or acked, we fill in the rate sample with the (prior)
+ * delivery information when the skb was last transmitted.
+ *
+ * If an ACK (s)acks multiple skbs (e.g., stretched-acks), this function is
+ * called multiple times. We favor the information from the most recently
+ * sent skb, i.e., the skb with the highest prior_delivered count.
+ */
+void tcp_rate_skb_delivered(struct sock *sk, struct sk_buff *skb,
+ struct rate_sample *rs)
+{
+ struct tcp_sock *tp = tcp_sk(sk);
+ struct tcp_skb_cb *scb = TCP_SKB_CB(skb);
+
+ if (!scb->tx.delivered_mstamp.v64)
+ return;
+
+ if (!rs->prior_delivered ||
+ after(scb->tx.delivered, rs->prior_delivered)) {
+ rs->prior_delivered = scb->tx.delivered;
+ rs->prior_mstamp = scb->tx.delivered_mstamp;
+ rs->is_retrans = scb->sacked & TCPCB_RETRANS;
+
+ /* Find the duration of the "send phase" of this window: */
+ rs->interval_us = skb_mstamp_us_delta(
+ &skb->skb_mstamp,
+ &scb->tx.first_tx_mstamp);
+
+ /* Record send time of most recently ACKed packet: */
+ tp->first_tx_mstamp = skb->skb_mstamp;
+ }
+ /* Mark off the skb delivered once it's sacked to avoid being
+ * used again when it's cumulatively acked. For acked packets
+ * we don't need to reset since it'll be freed soon.
+ */
+ if (scb->sacked & TCPCB_SACKED_ACKED)
+ scb->tx.delivered_mstamp.v64 = 0;
+}
+
+/* Update the connection delivery information and generate a rate sample. */
+void tcp_rate_gen(struct sock *sk, u32 delivered, u32 lost,
+ struct skb_mstamp *now, struct rate_sample *rs)
+{
+ struct tcp_sock *tp = tcp_sk(sk);
+ u32 snd_us, ack_us;
+
+ /* TODO: there are multiple places throughout tcp_ack() to get
+ * current time. Refactor the code using a new "tcp_acktag_state"
+ * to carry current time, flags, stats like "tcp_sacktag_state".
+ */
+ if (delivered)
+ tp->delivered_mstamp = *now;
+
+ rs->acked_sacked = delivered; /* freshly ACKed or SACKed */
+ rs->losses = lost; /* freshly marked lost */
+ /* Return an invalid sample if no timing information is available. */
+ if (!rs->prior_mstamp.v64) {
+ rs->delivered = -1;
+ rs->interval_us = -1;
+ return;
+ }
+ rs->delivered = tp->delivered - rs->prior_delivered;
+
+ /* Model sending data and receiving ACKs as separate pipeline phases
+ * for a window. Usually the ACK phase is longer, but with ACK
+ * compression the send phase can be longer. To be safe we use the
+ * longer phase.
+ */
+ snd_us = rs->interval_us; /* send phase */
+ ack_us = skb_mstamp_us_delta(now, &rs->prior_mstamp); /* ack phase */
+ rs->interval_us = max(snd_us, ack_us);
+
+ /* Normally we expect interval_us >= min-rtt.
+ * Note that rate may still be over-estimated when a spuriously
+ * retransmistted skb was first (s)acked because "interval_us"
+ * is under-estimated (up to an RTT). However continuously
+ * measuring the delivery rate during loss recovery is crucial
+ * for connections suffer heavy or prolonged losses.
+ */
+ if (unlikely(rs->interval_us < tcp_min_rtt(tp))) {
+ rs->interval_us = -1;
+ if (!rs->is_retrans)
+ pr_debug("tcp rate: %ld %d %u %u %u\n",
+ rs->interval_us, rs->delivered,
+ inet_csk(sk)->icsk_ca_state,
+ tp->rx_opt.sack_ok, tcp_min_rtt(tp));
+ }
+}