In SCTP, sending ACKs at the right time is presumably important in order
not to cause unnecessary retransmissions.
This CL sets the ACK timers to use high timer precision. This unblocks
experimentally lowering the precision of the default, "low", timers in
WebRTC.
// All bots are green, but mac_chromium_compile is randomly timing out
// independently of this CL and has been doing so for several days...
NOTRY=True
Bug: webrtc:13604
Change-Id: I1c81e3d0eeb477c3e00277d35649114c5edd249c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249090
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35813}
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}
This initial version contains a few tests, testing both lossy, non-lossy
and bandwidth limited networks. It uses simulated time, and runs much
faster than wall time on release builds, but slower on debug when there
is a lot of outstanding data (high throughput) as there are consistency
checks on outstanding data. Because of that, some tests had to be
disabled in debug build.
Bug: webrtc:12943
Change-Id: I9323f2dc99bca4e40558d05a283e99ce7dded7f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225201
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35124}
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}
This is mainly a refactoring commit, to break out packet sending to a
dedicated component.
Bug: webrtc:12943
Change-Id: I78f18933776518caf49737d3952bda97f19ef335
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228565
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34772}
It is now allowed in WebRTC, so let's use it.
Bug: webrtc:12943
Change-Id: I74a0f2fd9c1b9e7b5613ae1c592cf26842b8dddd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228564
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34768}
This is needed to be able to debug test cases when they fail.
Bug: webrtc:12961
Change-Id: I39bfe532709d02acb328ff5fdd005d33be4dc31c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225544
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34452}
The current send queue implements SCTP_SS_FCFS as defined in
https://datatracker.ietf.org/doc/html/rfc8260#section-3.1, but that has
always been known to be a temporary solution. The end goal is to
implement a Weighted Fair Queueing Scheduler (SCTP_SS_WFQ), but that's
likely to take some time.
Meanwhile, a round robin scheduler (SCTP_SS_RR) will be used to avoid
some issues with the current scheduler, such as a single data channel
completely blocking all others if it sends a lot of messages.
In this first commit, the code has simply been renamed and is still
implementing first-come-first-served. That will be fixed in follow-up
CLS.
Bug: webrtc:12793
Change-Id: Idc03b1594551bfe1ddbe1710872814b9fdf60cc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219684
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34090}
This should avoid the situation where WebRTC's GN check is green and
Chromium (which turns it ON for //third_party/webrtc) fails.
Bug: webrtc:12614
Change-Id: Id4c06ac57e9faa07c5e43491a61fbc093c68a40d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33900}
Those were found when trying to build within Chromium's codebase.
Bug: webrtc:12614
Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33896}
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}
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}
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}
It's responsible for answering incoming Heartbeat Requests, and to
send requests itself when a connection is idle. When it receives
a response, it will measure the RTT and if it doesn't receive a response
in time, that will result in a TX error, which will eventually close
the connection.
Bug: webrtc:12614
Change-Id: I08371d9072ff0461f60e0a2f7696c0fd7ccb57c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214129
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33828}
In the Socket module, there are a few (two, to be exact right now, but
the goal is to have even more) separate "handlers" that are responsible
for a feature set. These handlers must have an API to interact with
the rest of the socket - and this is the API.
Mocks are also added.
Bug: webrtc:12614
Change-Id: If19b43bf99a784bba3a42467d0ed3abdd8b4c62c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214128
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@{#33826}
