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}
The retransmission timeout (RTO) value is updated on every measured
RTT and is a function of the RTT value and its stability. In reality,
the RTT is never constant - it fluctuates, which makes the RTO become
much larger than the RTT. But for extremely stable RTTs, which we get
in simulations, the RTO value can become the same as the RTT, and that
makes expiration timers be scheduled to the RTT value, and will race
with packets that are expected to stop the expiration timer. And that
race should be avoided in simulations.
So ensuring that the RTO value is always greater, if only be a single
millisecond, will work fine in these simulations.
Bug: webrtc:12614
Change-Id: I30cf9c97e50449849ab35de52696c618d8498128
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219680
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34084}
Reporting the duplicate TSNs is a SHOULD in the RFC, and using the
duplicate TNSs is a MAY, and in reality I haven't seen an implementation
use it yet. However, it's good for debugging and for stats generation.
Bug: webrtc:12614
Change-Id: I1cc3f86961a8d289708cbf50d98dedfd25077955
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219462
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34053}
The minimum RTO time shouldn't be lower than the delayed ack timeout
of the peer to avoid sending retransmissions before the peer has
actually intended to reply.
In usrsctp, the default delayed ack timeout is 200ms and configurable
using the `sctp_delayed_sack_time_default` option. In dcsctp, it's
min(RTO/2, 200ms), to avoid this issue.
Bug: webrtc:12614
Change-Id: Ie84c331334af660d66b1a7d90d20f5cf7e2a5103
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219100
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34026}
The socket fuzzer is build as a structure-aware fuzzer where the full
public API is exercised as well as receival of SCTP packets with random
sequences of valid chunks.
It begins by putting the socket in a defined starting state and then,
based on the fuzzing data, performs a sequence of operations on the
socket such as receiving packets, sending data, resetting streams or
expiring timers.
This is the first iteration, and when running it a while and analyzing
code coverage, it will be modified to perform better. It could probably
be a little more random.
Bug: webrtc:12614
Change-Id: I50d6ffaecef5722be5cf666fee2f0de7d15cc2e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218500
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@{#33998}
When a socket is shutting down, either explicitly by the ULP calling
Shutdown(), or when the socket receives a SHUTDOWN chunk, the socket
should send all outstanding data and when all is sent and acked,
_then_ it should continue the shutdown protocol.
As it currently doesn't calculate correctly when all data has been sent,
as NACKED chunks are not included in what it believes is remaining in
the retransmission queue, it will shut down prematurely and may go back
to a previous state (ShutdownPending) from ShutdownSent or
ShutdownAckSent.
This is a workaround that just avoids the illegal state transition as
that puts the socket in an inconsistent state. The bug is merely
theoretical as WebRTC doesn't currently gracefully shut down a SCTP
socket, but just terminates the DTLS transport.
As TODOs mention, this will be fixed correctly a bit later.
Bug: webrtc:12739
Change-Id: Ibde2acc3a6aca701ac178d6181028404d470a5d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218340
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33982}
While it's not strictly defined, the expectation is that sending a
message with a lifetime parameter set to zero (0) ms should allow it to
be sent if it can be sent without being buffered. If it can't be
directly sent, it should be discarded.
This is initial support for it. Small messages can now be delivered fine
if they are not to be buffered, but fragmented messages could be partly
sent (if this fills up the congestion window), which means that the
message will then fail to be sent whenever the congestion window frees
up again. It would be better to - at a higher level - realize early that
the message can't be sent in full, and discard it without sending
anything. But that's an optimization that can be done later.
A few off-by-one errors were found when strictly defining that the
message is alive during its entire lifetime. It will expire just _after_
its lifetime.
Sending messages with a lifetime of zero may not supported in all
libraries, so a workaround would be to set a very small timeout instead,
which is tested as well.
Bug: webrtc:12614
Change-Id: I9a00bedb639ad7b3b565b750ef2a49c9020745f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217562
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33977}
Due to a limit socket send buffer, it's quite easy to fill it up when
using exponential slow start, which results in dropping a lot of packets
and having to retransmit those.
Disabling this, to align it to how SCTP normally behaves, and then try
to stabilize it later. With SCTP slow start, it will increase with one
MTU for each RTT when there is no packet loss. Even this mode will
experience packet loss, but not as much will be lost, and it will
stabilize quicker.
Bug: webrtc:12614
Change-Id: Ibc484b19b7e708fe5bd837bbef178a2f69b7211f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218203
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33969}
When the shutdown timer has expired, the socket will abort/close and the
TCB is not valid after InternalClose.
Bug: webrtc:12614
Change-Id: I09a94a049f0cda4577225dd9c80a92a8ec7e0423
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217767
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33956}
If Shutdown is called when the socket is being established and while the
connection timers are running, it will put the socket in an inconsistent
state, which is verified in debug builds.
Bug: webrtc:12614
Change-Id: I66f07d1170ac8f0ad9fd485d77d6aef4c365f150
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217765
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33949}
This makes it easier to understand which socket that experience an error
or abort. Aborts are now also logged, which was missed previously.
Bug: webrtc:12614
Change-Id: Ie5e4357b3e5450106cc6cc28c1e9578ad53d073a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217764
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33947}
This was caught in an integration test which had stricter assertions
than the FakeTimeout which is used in unit tests, so the first thing was
to add the same assertions to the FakeTimeout.
The issue is that when a Timer triggers, and if it's set to
automatically restart (possibly with an exponential backoff), the
`is_running_` field was set to true while the timer callback was called
to allow the client to know that the timer is in fact running, but the
timer was actually not started until the callback returned. Which made
sense, as the callback can with its return value override the duration,
which should affect the backoff algorithm.
The problem was when a timer was manually started within the callback.
As the Timer itself thought that it was already running, it first would
Stop() the underlying Timeout, then Start(). But calling Stop() on a
timeout that is not running is illegal, which set of assertions.
So the solution is to don't lie; Don't say that a timer is running when
it's not. Make sure that the timer is running when the timer callback is
triggered, which makes it consistent at all times. That may result in
unnecessary timeout invocations (stopping and starting), but that's not
too expensive.
Bug: webrtc:12614
Change-Id: I7b4447ccd88bd43d181e158f0d29b0770c8a3fd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217522
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33926}
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}
As INFO is aliased to LS_INFO, this didn't trigger any warnings or
compilation errors.
Bug: None
Change-Id: I1ed30c435d9ee6ea1b51d85a375d70135d3475e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216689
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33912}
This is about doing the best with what we have. As delayed tasks can't
be cancelled, and dcSCTP timers will almost always be stopped or
restarted, and will generally only expire on packet loss.
This implementation will post a delayed task whenever a Timeout is
started. Whenever it's stopped or restarted, it will keep the scheduled
delay task running (there's no alternative), but it will also not start
a new delayed task on subsequent starts/restarts. Instead, it will wait
until the original delayed task has triggered, and will then - if the
timer is still running, which it probably isn't - post a new delayed
task with the remainder of the the duration.
There is special handling for when a shorter duration is requested, as
that can't re-use the scheduled task, but that shouldn't be very common.
Bug: webrtc:12614
Change-Id: I7f3269cabf84f80dae3b8a528243414a93d50fc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217223
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33904}
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}
Ensuring that timer durations never go beyond a safe maximum duration
and that timer IDs are not re-used.
Bug: webrtc:12614
Change-Id: I227a2e9933da16669dc6ea0a39c570892010ba2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215063
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33860}
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}
The Retransmission Queue contain all message fragments (DATA chunks)
that have once been sent, but not yet ACKed by the receiver. It will
process incoming SACK chunks, which informs it which chunks that the
receiver has seen (ACKed) and which that are lost (NACKed), and will
retransmit chunks when it's time.
If a message has been sent with partial reliability, e.g. to have a
limited number of retransmissions or a limited lifetime, the
Retransmission Queue may discard a partially sent and expired message
and will instruct the receiver that "don't expect this message - it's
expired" by sending a FORWARD-TSN chunk.
This currently also includes the congestion control algorithm as it's
tightly coupled with the state of the retransmission queue. This is
a fairly complicated piece of logic which decides how much data that
can be in-flight, depending on the available bandwidth. This is not done
by any bandwidth estimation, but similar to TCP, where data is sent
until it's lost, and then "we dial down a knob" and take it more
carefully from here on.
Future refactoring will try to separate the logic regarding fragment
retransmission and the congestion control algorithm.
Bug: webrtc:12614
Change-Id: I8678250abb766e567c3450634686919936ea077b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214046
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33833}
The socket can measure the round-trip-time (RTT) by two different
scenarios:
* When a sent data is ACKed
* When a HEARTBEAT has been sent, which as been ACKed.
The RTT will be used to calculate which timeout value that should be
used for e.g. the retransmission timer (T3-RTX). On connections with a
low RTT, the RTO value will be low, and on a connection with high RTT,
the RTO value will be high. And on a connection with a generally low
RTT value, but where it varies a lot, the RTO value will be calculated
to be fairly high, to not fire unnecessarily. So jitter is bad, and is
part of the calculation.
Bug: webrtc:12614
Change-Id: I64905ad566d5032d0428cd84143a9397355bbe9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214045
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33832}
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}
This is just a simple SCTP variable, but wrapped in its own object
for convenience.
Bug: webrtc:12614
Change-Id: I0c45c356488d21b71c72a936e4ceeee5ed0ec96d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214047
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33827}
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}
When the client asks for a message to be sent, it's put in the
SendQueue, which is available even when the socket is not yet connected.
When the socket is connected, those messages will be sent on the wire,
possibly fragmented if the message is large enough to not fit inside a
single packet. When the message has been fully sent, it's removed from
the send queue (but it will be in the RetransmissionQueue - which is
added in a follow-up change, until the message has been ACKed).
The Send Queue is a FIFO queue in this iteration, and in SCTP, that's
called a "First Come, First Served" queue, or FCFS. In follow-up work,
the queue and the actual scheduling algorithm which decides which
message that is sent, when there are messages in multiple streams, will
likely be decoupled. But in this iteration, they're in the same class.
Bug: webrtc:12614
Change-Id: Iec1183e625499a21e402e4f2a5ebcf989bc5c3ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214044
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33798}
Adding strong types went a little too far as these two types represent
the same sequence number. A "request sequence number" is a number, that
- when responded to - will be used as "response sequence number".
Having them separate added confusion and just a lot of type-casting.
Bug: webrtc:12614
Change-Id: I4636ea8f2252023a2d5a9b7033763e1978b1812e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214130
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33789}
The SCTP RFCs aren't very strict in specifying when a chunk or parameter
is invalid, so most chunks and/or parameters must be accepted but they
may need some cleaning to avoid a lot of error handling deeper in the
chunk handling code.
Bug: webrtc:12614
Change-Id: I723f08cbdc26e1a1b78463b6137340e638089037
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214966
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33788}
This test verifies that a StrongAlias<bool> can be evaluated as
a boolean without dereferencing it. Note that this is not the case
for StrongAlias<int>, for example, as that wouldn't even compile. Which
is quite good.
Bug: webrtc:12614
Change-Id: I67329364721fe0354d78daac1233254035454c03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215686
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33787}
Due to a previous refactoring, the SCTP packet header is only added when
the first chunk is written. This wasn't reflected in the
`bytes_remaining`, which made it add more than could fit within the MTU.
Additionally, the maximum packet size must be even divisible by four as
padding will be added to chunks that are not even divisble by four (up
to three bytes of padding). So compensate for that.
Bug: webrtc:12614
Change-Id: I6b57dfbf88d1fcfcbf443038915dd180e796191a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215145
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33760}
To be able to use them type-safely, they should support native
operators (e.g. adding a time and a duration, or subtracting two time
values), as the alternative is to manage them as numbers.
Yes, this makes them behave a bit like absl::Time/absl::Duration.
Bug: webrtc:12614
Change-Id: I4dea12e33698a46e71fb549f44c06f2f381c9201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215143
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33725}
These are some fixes that were added after submission of
https://webrtc-review.googlesource.com/c/src/+/213664
Mainly:
* Don't accept TSNs that have a too large difference from expected
* Renaming of member variable (to confirm to style guidelines)
Bug: webrtc:12614
Change-Id: I06e11ab2acf5d307b68c3cbc135fde2c038ee690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215070
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33721}
* `AddTo` and `Difference` are made into static methods, as one may have
believed that these modified the current object previously. The
`Increment` method is kept, as it's obvious that it modifies the
current object as it doesn't have a return value, and `next_value` is
kept, as its naming (lower-case, snake) indicates that it's a simple
accessor.
* Difference will return the absolute difference. This is actually the
only reasonable choice, as the return value was unsigned and any
negative value would just wrap.
Bug: webrtc:12614
Change-Id: If14a71636e67fc612d12759dc80a9c2518c85281
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215069
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33714}
Adding fuzzers to the build made "gn gen --check" discover a lot
of dependency errors between various components of dcSCTP.
Bug: webrtc:12614
Change-Id: I0b2dd7321aec2624da417f413c727bd11b4743e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215003
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33705}
When migrating to use StrongAlias types, the PPID was incorrectly
modeled as an uint16_t instead of a uint32_t, as it was prior to using
StrongAlias. Most likely a copy-paste error from StreamID.
As the Data Channel PPIDs are in the range of 51-57, it was never caught
in tests.
Bug: webrtc:12614
Change-Id: I2b61ef7935df1222068e7f4e70fc2aaa532dcf7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214960
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33687}
The Reassembly Queue receives fragmented messages (DATA or I-DATA
chunks) and - with help of stream reassemblers - will reassemble these
fragments into messages, which will be delivered to the client.
It also handle partial reliability (FORWARD-TSN) and stream resetting.
To avoid a DoS attack vector, where a sender can send fragments in a way
that the reassembly queue will never succeed to reassemble a message and
use all available memory, the ReassemblyQueue has a maximum size.
Bug: webrtc:12614
Change-Id: Ibb084fecd240d4c414e096579244f8f5ee46914e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214043
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33678}
This class handles the assembly of fragmented received messages (as DATA
chunks) and manage per-stream queues. This class only handles
non-interleaved messages as described in RFC4960, and is not used when
message interleaving is enabled on the association, as described in
RFC8260.
This is also only part of the reassembly - a follow-up change will add
the ReassemblyQueue that handle the other part as well. And an even
further follow-up change will add a "interleaved reassembly stream".
Bug: webrtc:12614
Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33677}
The Data Tracker's purpose is to keep track of all received DATA chunks
and to ACK/NACK that data, by generating SACK chunks reflecting its view
of what has been received and what has been lost.
It also contains logic for _when_ to send the SACKs, as that's different
depending on e.g. packet loss. Generally, SACKs are sent every second
packet on a connection with no packet loss, and can also be sent on a
delayed timer.
In case partial reliability is used, and the transmitter has decided
that some data shouldn't be retransmitted, it will send a FORWARD-TSN
chunk, which this class also handles, by "forgetting" about those
chunks.
Bug: webrtc:12614
Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33676}
