tree 998c5e15fbab90e6cad7cef4c2a712452cdc2d3e
parent a9e7f719a90c1663a498177f71c30eb9e5b11c68
author Victor Boivie <boivie@webrtc.org> 1628707427 +0200
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> 1628788975 +0000

dcsctp: Add burst limiter for sent packets

Some deployments, e.g. Chromium, has a limited send buffer. It's
reasonable that it's quite small, as it avoids queuing too much, which
typically results in increased latency for real-time communication. To
avoid SCTP to fill up the entire buffer at once - especially when doing
fast retransmissions - limit the amount of packets that are sent in one
go.

In a typical scenario, SCTP will not send more than three packets for
each incoming packet, which is is the case when a SACK is received which
has acknowledged two large packets, and which also adds the MTU to the
congestion window (due to in slow-start mode), which then may result in
sending three packets. So setting this value to four makes any
retransmission not use that much more of the send buffer.

This is analogous to usrsctp_sysctl_set_sctp_fr_max_burst_default in
usrsctp, which also has the default value of four (4).

Bug: webrtc:12943
Change-Id: Iff76a1668beadc8776fab10312ef9ee26f24e442
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34744}