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The socket can measure the round-trip-time (RTT) by two different scenarios: * When a sent data is ACKed * When a HEARTBEAT has been sent, which as been ACKed. The RTT will be used to calculate which timeout value that should be used for e.g. the retransmission timer (T3-RTX). On connections with a low RTT, the RTO value will be low, and on a connection with high RTT, the RTO value will be high. And on a connection with a generally low RTT value, but where it varies a lot, the RTO value will be calculated to be fairly high, to not fire unnecessarily. So jitter is bad, and is part of the calculation. Bug: webrtc:12614 Change-Id: I64905ad566d5032d0428cd84143a9397355bbe9f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214045 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33832} |
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| .. | ||
| BUILD.gn | ||
| dcsctp_message.h | ||
| dcsctp_options.h | ||
| dcsctp_socket.h | ||
| packet_observer.h | ||
| strong_alias.h | ||
| strong_alias_test.cc | ||
| timeout.h | ||
| types.h | ||
| types_test.cc | ||