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* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <cstdint>
#include <functional>
#include "net/dcsctp/public/dcsctp_options.h"
namespace dcsctp {
// Manages updating of the Retransmission Timeout (RTO) SCTP variable, which is
// used directly as the base timeout for T3-RTX and for other timers, such as
// delayed ack.
// When a round-trip-time (RTT) is calculated (outside this class), `Observe`
// is called, which calculates the retransmission timeout (RTO) value. The RTO
// value will become larger if the RTT is high and/or the RTT values are varying
// a lot, which is an indicator of a bad connection.
class RetransmissionTimeout {
explicit RetransmissionTimeout(const DcSctpOptions& options);
// To be called when a RTT has been measured, to update the RTO value.
void ObserveRTT(DurationMs measured_rtt);
// Returns the Retransmission Timeout (RTO) value, in milliseconds.
DurationMs rto() const { return DurationMs(rto_); }
// Returns the smoothed RTT value, in milliseconds.
DurationMs srtt() const { return DurationMs(srtt_); }
// Note that all intermediate state calculation is done in the floating point
// domain, to maintain precision.
const double min_rto_;
const double max_rto_;
const double max_rtt_;
// If this is the first measurement
bool first_measurement_ = true;
// Smoothed Round-Trip Time
double srtt_ = 0.0;
// Round-Trip Time Variation
double rttvar_ = 0.0;
// Retransmission Timeout
double rto_;
} // namespace dcsctp