10 Commits

Author SHA1 Message Date
Henrik Boström
b951dc6f4c Allow specifying delayed task precision of dcsctp::Timer.
Context: The timer precision of PostDelayedTask() is about to be lowered
to include up to 17 ms leeway. In order not to break use cases that
require high precision timers, PostDelayedHighPrecisionTask() will
continue to have the same precision that PostDelayedTask() has today.
webrtc::TaskQueueBase has an enum (kLow, kHigh) to decide which
precision to use when calling PostDelayedTaskWithPrecision().

See go/postdelayedtask-precision-in-webrtc for motivation and a table of
delayed task use cases in WebRTC that are "high" or "low" precision.

Most timers in DCSCTP are believed to only be needing low precision (see
table), but the delayed_ack_timer_ of DataTracker[1] is an example of a
use case that is likely to break if the timer precision is lowered (if
ACK is sent too late, retransmissions may occur). So this is considered
a high precision use case.

This CL makes it possible to specify the precision of dcsctp::Timer.
In a follow-up CL we will update delayed_ack_timer_ to kHigh precision.

[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/net/dcsctp/rx/data_tracker.cc;l=340

Bug: webrtc:13604
Change-Id: I8eec5ce37044096978b5dd1985fbb00bc0d8fb7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249081
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35809}
2022-01-26 18:40:24 +00:00
Victor Boivie
f4fa166cc5 dcsctp: Detect the peer SCTP implementation
It's to be used for clients to record metrics and to e.g. attribute
metrics to which SCTP implementation the peer was using.

This is not explicitly signaled, so heuristics are used. These are not
guaranteed to come to the correct conclusion, and the data is not always
available.

Note: The behavior of dcSCTP will not change depending on the assumed
implementation - only by explicitly signaled capabilities.

Bug: webrtc:13216
Change-Id: I2f58054d17d53d947ed5845df7a08f974d42f918
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35103}
2021-09-28 05:10:45 +00:00
Sergey Sukhanov
4397281f38 dcsctp: implement socket handover in the DcSctpSocket class and expose the functionality in the API
Bug: webrtc:13154
Change-Id: Idf4f4028c8e65943cb6b41fab0baef1b3584205d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232126
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35029}
2021-09-17 15:19:01 +00:00
Sergey Sukhanov
923d2c237e dcsctp: fixed grammar in one comment, added comment regarding the threading contract
Bug: None
Change-Id: Ia1442a155afb38b0df4ed2c288a9c6b238488b23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229080
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/master@{#34788}
2021-08-17 15:59:05 +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
d4716eaf60 dcsctp: Add metrics support
To support implementing RTCSctpTransportStats, a few metrics are needed.

Some more were added that are useful for metric collection in SFUs.

Bug: webrtc:13052
Change-Id: Idafd49e1084922d01d3e6c5860715f63aea08b7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228243
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34708}
2021-08-10 20:01:46 +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
Florent Castelli
0810b05104 dcsctp: Add SetMaxMessageSize() to socket
An SCTP transport for Data Channels allows changing the maximum
message size through SDP.
See https://w3c.github.io/webrtc-pc/#sctp-transport-update-mms

Bug: webrtc:12614
Change-Id: I8cff33c5f9c1d60934a726c546bc9cbdcd9e22d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217387
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33920}
2021-05-04 21:43:24 +00:00
Victor Boivie
b6580ccb29 dcsctp: Add Socket
This completes the basic implementation of the dcSCTP library. There
are a few remaining commits to e.g. add compatibility tests and
benchmarks, as well as more support for e.g. RFC8260, but those are not
strictly vital for evaluation of the library.

The Socket contains the connection establishment and teardown sequences
as well as the general chunk dispatcher.

Bug: webrtc:12614
Change-Id: I313b6c8f4accc144e3bb88ddba22269ebb8eb3cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214342
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33890}
2021-05-01 07:16:21 +00:00
Victor Boivie
628d91cd0d dcsctp: Add public API
Clients will use this API for all their interactions with this library.
It's made into an interface (of which there will only be a single
implementation) simply for documentation purposes. But that also allows
clients to mock the library if wanted or to have a thread-safe wrapper,
as the library itself is not thread-safe, while keeping the same
interface.

Bug: webrtc:12614
Change-Id: I346af9916e26487da040c01825c2547aa7a5d0b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213348
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33648}
2021-04-08 08:53:44 +00:00