Reducing threshold for usrsctp "buffer low" callback.
A usrsctp regression is causing this callback to not be invoked, but
reducing the threshold (from 128KB to 64KB) seems to mitigate the issue.
Can set it back once the root cause is fixed, though this isn't
expected to have any performance implications.
Bug: webrtc:11824
Change-Id: I2f6a3183d298abf4d1ad3bbd3697b1879eb4d696
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180841
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31851}
diff --git a/media/sctp/sctp_transport.cc b/media/sctp/sctp_transport.cc
index 8996288..9fe8717 100644
--- a/media/sctp/sctp_transport.cc
+++ b/media/sctp/sctp_transport.cc
@@ -781,7 +781,10 @@
// If kSctpSendBufferSize isn't reflective of reality, we log an error, but we
// still have to do something reasonable here. Look up what the buffer's real
// size is and set our threshold to something reasonable.
- static const int kSendThreshold = usrsctp_sysctl_get_sctp_sendspace() / 2;
+ // TODO(bugs.webrtc.org/11824): That was previously set to 50%, not 25%, but
+ // it was reduced to a recent usrsctp regression. Can return to 50% when the
+ // root cause is fixed.
+ static const int kSendThreshold = usrsctp_sysctl_get_sctp_sendspace() / 4;
sock_ = usrsctp_socket(
AF_CONN, SOCK_STREAM, IPPROTO_SCTP, &UsrSctpWrapper::OnSctpInboundPacket,
