rustdesk e50d620889 webrtc: take dcsctp's retransmission timings and IPv6-safe MTU ...

webrtc-sctp ships RFC 4960's RTO.Initial/RTO.Min (3000/1000), TCP's values for
arbitrary public paths. On this workload they set the recovery time outright:
a request/response exchange keeps one chunk in flight, so no later SACK ever
raises miss_indicator to the 3 that arms fast retransmit, and the T3 floor is
the only way back. A single loss during a handshake or a first keyframe
therefore costs whole seconds.

The fork now carries dcsctp's numbers instead - the SCTP implementation Google
wrote to replace usrsctp for Chrome's WebRTC data channels, the same realtime
workload: rto_initial 500, rto_min 400, a 220ms floor under the RTT variance,
and mtu 1191. INITIAL_MTU 1228 plus DTLS/UDP/IPv6 overhead is 1313, past the
1280 minimum, so every full-size chunk fragmented on an IPv6 path.

Both patch entries move to the new branch, which also carries the Windows IPv6
byte-swap fix, so one rev matches the whole webrtc 0.13 stack.

2026-08-27 14:49:34 +08:00

263 KiB

Raw Blame History

View Raw