Change initial DTLS retransmission timer from 1 second to 50ms.
This will help ensure a timely DTLS handshake when there's packet
loss. It will likely result in spurious retransmissions (since the
RTT is usually > 50ms), but since exponential backoff is still used,
there will at most be ~4 extra retransmissions. For a time-sensitive
application like WebRTC this seems like a reasonable tradeoff.
R=pthatcher@webrtc.org, juberti@chromium.org, juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1981463002 .
Committed: https://crrev.com/1e435628366fb9fed71632369f05928ed857d8ef
Cr-Original-Commit-Position: refs/heads/master@{#12853}
Cr-Commit-Position: refs/heads/master@{#13159}
diff --git a/webrtc/base/logging.cc b/webrtc/base/logging.cc
index 6060362..c951e15 100644
--- a/webrtc/base/logging.cc
+++ b/webrtc/base/logging.cc
@@ -124,7 +124,9 @@
const char* module)
: severity_(sev), tag_(kLibjingle) {
if (timestamp_) {
- int64_t time = TimeSince(LogStartTime());
+ // Use SystemTimeMillis so that even if tests use fake clocks, the timestamp
+ // in log messages represents the real system time.
+ int64_t time = TimeDiff(SystemTimeMillis(), LogStartTime());
// Also ensure WallClockStartTime is initialized, so that it matches
// LogStartTime.
WallClockStartTime();
@@ -210,7 +212,7 @@
}
int64_t LogMessage::LogStartTime() {
- static const int64_t g_start = TimeMillis();
+ static const int64_t g_start = SystemTimeMillis();
return g_start;
}