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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 "net/dcsctp/tx/retransmission_timeout.h"
#include <algorithm>
#include <cstdint>
#include "net/dcsctp/public/dcsctp_options.h"
namespace dcsctp {
RetransmissionTimeout::RetransmissionTimeout(const DcSctpOptions& options)
: min_rto_(*options.rto_min),
max_rto_(*options.rto_max),
max_rtt_(*options.rtt_max),
min_rtt_variance_(*options.min_rtt_variance),
scaled_srtt_(*options.rto_initial << kRttShift),
rto_(*options.rto_initial) {}
void RetransmissionTimeout::ObserveRTT(DurationMs measured_rtt) {
const int32_t rtt = *measured_rtt;
// Unrealistic values will be skipped. If a wrongly measured (or otherwise
// corrupt) value was processed, it could change the state in a way that would
// take a very long time to recover.
if (rtt < 0 || rtt > max_rtt_) {
return;
}
// From https://tools.ietf.org/html/rfc4960#section-6.3.1, but avoiding
// floating point math by implementing algorithm from "V. Jacobson: Congestion
// avoidance and control", but adapted for SCTP.
if (first_measurement_) {
scaled_srtt_ = rtt << kRttShift;
scaled_rtt_var_ = (rtt / 2) << kRttVarShift;
first_measurement_ = false;
} else {
int32_t rtt_diff = rtt - (scaled_srtt_ >> kRttShift);
scaled_srtt_ += rtt_diff;
if (rtt_diff < 0) {
rtt_diff = -rtt_diff;
}
rtt_diff -= (scaled_rtt_var_ >> kRttVarShift);
scaled_rtt_var_ += rtt_diff;
}
if (scaled_rtt_var_ < min_rtt_variance_) {
scaled_rtt_var_ = min_rtt_variance_;
}
rto_ = (scaled_srtt_ >> kRttShift) + scaled_rtt_var_;
// Clamp RTO between min and max.
rto_ = std::min(std::max(rto_, min_rto_), max_rto_);
}
} // namespace dcsctp
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