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}
27e50ccf4c