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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}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.