Commit graph

6 commits

Author SHA1 Message Date
Sergey Sukhanov
3852698ad9 dcsctp: support handover state serialization testing
dcSCTP library users can set their custom
g_handover_state_transformer_for_test that can serialize and
deserialize the state. All dcSCTP handover tests call
g_handover_state_transformer_for_test. If some part of the state is
serialized incorrectly or is forgotten, high chance that it will
fail the tests.

Bug: webrtc:13154
Change-Id: I251a099be04dda7611e9df868d36e3a76dc7d1e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232325
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35035}
2021-09-20 10:08:58 +00:00
Sergey Sukhanov
3b08fe3dcd dssctp: support socket handover in StreamResetHandler
Bug: webrtc:13154
Change-Id: Idafbed4f3c1af8d0cca833ba983c4b4b99118335
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232121
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35012}
2021-09-16 11:33:21 +00:00
Victor Boivie
8df32eb0e1 dcsctp: Add API to indicate packet send status
Before this change, there was no way for a client to indicate to the
dcSCTP library if a packet that was supposed to be sent, was actually
sent. It was assumed that it always was.

To handle temporary failures better, such as retrying to send packets
that failed to be sent when the send buffer was full, this information
is propagated to the library.

Note that this change only covers the API and adaptations to clients.
The actual implementation to make use of this information is done as a
follow-up change.

Bug: webrtc:12943
Change-Id: I8f9c62e17f1de1566fa6b0f13a57a3db9f4e7684
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228563
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34767}
2021-08-16 11:29:47 +00:00
Victor Boivie
236ac50628 dcsctp: Add public API for BufferedAmountLow
This adds native support for the RTCDataChannel properties:
https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel/bufferedAmount
https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel/bufferedAmountLowThreshold

And the RTCDataChannel event:
https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel/onbufferedamountlow

The old callback, NotifyOutgoingMessageBufferEmpty, is deprecated as it
didn't work very well. It will not be triggered and will be removed
as soon as all users of it are gone. There is a new callback,
OnTotalBufferedAmountLow, that serves the same purpose but also allows
setting an arbitrary limit when it should be triggered (See
DcSctpOptions::total_buffered_amount_low_threshold).

Bug: webrtc:12794
Change-Id: Ic1c92f174eff8a1acda0b5fd3dcc45bd1cfa2704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219691
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34144}
2021-05-27 15:27:27 +00:00
Victor Boivie
1d2fa9a1c3 dcsctp: Expire timers just before triggering them
In real life, when a Timeout expires, the caller is supposed to call
DcSctpSocket::HandleTimeout directly, as the Timeout that just expired
is stopped (it just expired), but the Timer still believes it's running.
The system is not in a consistent state.

In tests, all timeouts were evaluated at the same time, which, if two
timeouts expired at the same time, would put them both as "not running",
and with their timers believing they were running. So if you would do
any operation on a timer whose timeout had just expired, the timeout
would assert saying that "you can't stop a stopped timeout" or similar.

This isn't relevant in non-test scenarios.

Solved by expiring timeouts one by one.

Bug: webrtc:12614
Change-Id: I79d006f4d3e96854d77cec3eb0080aa23b8569cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217560
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33925}
2021-05-05 12:40:21 +00:00
Victor Boivie
322f911c10 dcsctp: Add Stream Reset Handler
The Stream Reset handler handles a limited subset of RFC6525, but all
the parts necessary to implement "Closing a Data Channel", which is done
by sending an Outgoing SSN Reset Request.

There can only be a single "Stream Reconfiguration Request" on the wire
at any time, so requests are queued and sent when a previous request -
if any - finishes. Resetting a stream is an asynchronous operation and
the receiver will not perform the stream resetting until it can be done,
which is when the currently partly received message has been fully
received. And the sender will not send a request until the currently
fragmented message (on that stream) is still sent.

There are numerous callbacks to make the client know what's really
happening as these callbacks will result in Data Channel events.

Bug: webrtc:12614
Change-Id: I9fd0a94713f0c1fc384d1189f3894e87687408b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214131
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33856}
2021-04-27 21:08:56 +00:00