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 CL adds the API to enable message lifecycle events to be generated.
Those can in turn be used to generate metrics, e.g. latency metrics
tracking the time to send a message, the time until it's acknowledged,
and metrics tracking how often messages are expired.
This will be used to validate that message interleaving really improves
latency for high priority data channels.
The actual implementation of the API will be provided in follow-up CLs.
Bug: webrtc:5696
Change-Id: Ic06f8244d1c79a336975e35479130521dff17519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264141
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37396}
There was also some refactoring to create the TCB at the same time,
to ensure the metric is always set.
Bug: webrtc:13052, webrtc:5696
Change-Id: I5557ad5f0fc4a0520de1eaaafa15459b3200c4f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262259
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37388}
This is a reland of commit 17a02a31d7.
This is the first part of supporting stream priorities, and adds the API
and very basic support for setting and retrieving the stream priority.
This commit doesn't in any way change the actual packet sending - the
specified priority values are stored, but not acted on.
This is all that is client visible, so clients can start using the API
as written, and they would never notice that things are missing.
Bug: webrtc:5696
Change-Id: I04d64a63cbaec67568496ad99667e14eba85f2e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264424
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37081}
This reverts commit 17a02a31d7.
Reason for revert: Breaks downstream test
Original change's description:
> dcsctp: Add public API for setting priorities
>
> This is the first part of supporting stream priorities, and adds the API
> and very basic support for setting and retrieving the stream priority.
>
> This commit doesn't in any way change the actual packet sending - the
> specified priority values are stored, but not acted on.
>
> This is all that is client visible, so clients can start using the API
> as written, and they would never notice that things are missing.
>
> Bug: webrtc:5696
> Change-Id: I24fce8cbb6f3cba187df99d1d3f45e73621c93c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261943
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Victor Boivie <boivie@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37034}
Bug: webrtc:5696
Change-Id: If172d9c9dbce7aae72152abbbae1ccc77340bbc1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264444
Owners-Override: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37039}
This is the first part of supporting stream priorities, and adds the API
and very basic support for setting and retrieving the stream priority.
This commit doesn't in any way change the actual packet sending - the
specified priority values are stored, but not acted on.
This is all that is client visible, so clients can start using the API
as written, and they would never notice that things are missing.
Bug: webrtc:5696
Change-Id: I24fce8cbb6f3cba187df99d1d3f45e73621c93c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261943
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37034}
This CL first restricts Metrics to be retrievable when the socket is
created. This avoids having most fields as optional and makes it
easier to add more metrics.
Secondly, the peer implementation is moved into Metrics.
Bug: webrtc:13052
Change-Id: I6cb53eeef3f84ac34f3efc883853338f903cc758
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262256
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36888}
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}
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}
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}
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}
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}
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}
