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#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.
*
* TCP flows can often be application-limited in request/response workloads.
* The estimator marks a bandwidth sample as application-limited if there
* was some moment during the sampled window of packets when there was no data
* ready to send in the write queue.
*/
/* 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;
TCP_SKB_CB(skb)->tx.is_app_limited = tp->app_limited ? 1 : 0;
}
/* 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_app_limited = scb->tx.is_app_limited;
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,
bool is_sack_reneg, struct skb_mstamp *now, struct rate_sample *rs)
{
struct tcp_sock *tp = tcp_sk(sk);
u32 snd_us, ack_us;
/* Clear app limited if bubble is acked and gone. */
if (tp->app_limited && after(tp->delivered, tp->app_limited))
tp->app_limited = 0;
/* 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 or
* in recovery from loss with SACK reneging. Rate samples taken during
* a SACK reneging event may overestimate bw by including packets that
* were SACKed before the reneg.
*/
if (!rs->prior_mstamp.v64 || is_sack_reneg) {
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))) {
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));
rs->interval_us = -1;
return;
}
/* Record the last non-app-limited or the highest app-limited bw */
if (!rs->is_app_limited ||
((u64)rs->delivered * tp->rate_interval_us >=
(u64)tp->rate_delivered * rs->interval_us)) {
tp->rate_delivered = rs->delivered;
tp->rate_interval_us = rs->interval_us;
tp->rate_app_limited = rs->is_app_limited;
}
}
/* If a gap is detected between sends, mark the socket application-limited. */
void tcp_rate_check_app_limited(struct sock *sk)
{
struct tcp_sock *tp = tcp_sk(sk);
if (/* We have less than one packet to send. */
tp->write_seq - tp->snd_nxt < tp->mss_cache &&
/* Nothing in sending host's qdisc queues or NIC tx queue. */
sk_wmem_alloc_get(sk) < SKB_TRUESIZE(1) &&
/* We are not limited by CWND. */
tcp_packets_in_flight(tp) < tp->snd_cwnd &&
/* All lost packets have been retransmitted. */
tp->lost_out <= tp->retrans_out)
tp->app_limited =
(tp->delivered + tcp_packets_in_flight(tp)) ? : 1;
}
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