https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-zero-checksum/06/
The previous implementation was for version 00, and since then changes
have been made. The chunk that is used to negotiate this capability has
now grown to include an additional property - the sender's alternate
error detection method.
Bug: webrtc:14997
Change-Id: I78043d187b79f40bbadbcba02eae6eedf54f30f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336380
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41759}
The handover state has been added with
commit daaa6ab5a8.
This reverts commit 014cbed9d2.
Bug: webrtc:14997
Change-Id: Ie84f3184f3ea67aaa6438481634046ba18b497a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40794}
This reverts commit a736f30a5f.
Due to a downstream project not supporting this
new handover state, it fails. Handover state needs
to be submitted first, then downstream project needs
to be updated, and finally the code changes can be
submitted.
Bug: webrtc:14997
Change-Id: I8551e349408d9bf4eb593cb948279d659467fe20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302821
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39923}
If configured, attempt to negotiate "zero checksum
acceptable" capability, which will make the outgoing
packets have a fixed checksum of zero. Received
packets will not be verified for a correct checksum
if it's zero.
Also includes some boilerplate:
- Setting capability in state cookie
- Adding capability to handover state
- Adding metric to know if the feature is used
This feature is not enabled by default, as it will be
first evaluated with an A/B experiment before making
it the default.
When the feature is enabled, lower CPU consumption for
both receiving and sending packets is expected. How
much depends on the architecture, as some architectures
have better support for generating CRC32 in hardware
than others.
Bug: webrtc:14997
Change-Id: If23f73e87220c7d42bd4f9a92772cda16bc18fcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299076
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39920}
To allow the transport to be able to know which ranges of
stream identifiers it can use, the negotiated incoming/inbound
and outgoing/outbound stream counts will be exposed. They are
added to Metrics, and guaranteed to be available from within
the OnConnected callback.
In this CL, dcSCTP will not validate that the client is sending
on a stream that is within the negotiated bounds. That will be
done as a follow-up CL.
Bug: webrtc:14277
Change-Id: Ic764e5f93f53d55633ee547df86246022f4097cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272321
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37876}
This is merely a container of components that have their lifetime
bound to when the socket is connected. If the socket gets disconnected
or restarted, this object (and everything it holds) will be released.
Bug: webrtc:12614
Change-Id: Ibd75760b7bf7efe9c26c4eb7cee62de8bba5410c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214340
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33869}
