This avoids a couple of layers of error code conversion, reduces
dependency on cricket error types and allows us to preserve error
information from dcsctp. Along the way remove SendDataResult.
Bug: none
Change-Id: I1ad18a8f0b2fb181745b19c49f36f270708720c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298305
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39619}
This struct only contains two member variables now and there isn't
much value added by having it.
Low-Coverage-Reason: No change in coverage, CL modifies uncovered RTC_LOG lines.
Bug: none
Change-Id: I924d450f4c8f8e49b1cfeabaebee9fd5235a90cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297360
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39563}
Instead there are direct member variables for the various relevant
states, some weren't needed, some can be const but the `id` member
in particular needs special handling and can't be const.
For dealing with the stream id, we now have SctpSid. A class that does range validation, checks thread safety, handles the special `-1` case (for what's essentially an unsigned 16 bit int). Using a special type
for this also has the effect that range checking happens more
consistently (although I'm not modifying the structs in api/).
With upcoming steps of avoiding thread hops, the ID may need to
migrate to the network thread, which the thread checks will help with.
Along the way, update SctpSidAllocator to use flat_set instead of std::set and moving some of the sctp data channel code to the cc file
to help with more accurately tracking code coverage.
Bug: webrtc:11547
Change-Id: Iea6e7647ab8f93052044c5afbcc449115206b4e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296444
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39539}
In rare cases, it is possible to queue a call to SendData from the signaling
thread on a channel being closed or already closed in the network thread.
By keeping track of currently open streams, we avoid sending messages
with a stream id of channels that the other side already considers closed
and has already reused for a new channel.
This caused rare messages to be delivered on the wrong data channel if
a message was quickly sent, channel closed and a new one reopened.
Bug: webrtc:14277
Change-Id: If35fed8d12d5d2c18cdc6601085d8b632c37a0ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272624
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37880}
The state machine for handling resets couldn't handle resets
happening from both sides at the same time.
Bug: webrtc:13994
Change-Id: I2c268e54f4c5c9858913faef91ff00f6af956e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261305
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36799}
When an SCTP stream is closing, a stream reset needs
to be sent from both ends.
The remote was not sending a stream reset and quickly
opening another stream with the same StreamID could
cause SCTP errors.
Bug: webrtc:13994
Change-Id: I3abc74ddc88b3fcf7e6495d76e7d77f52280b5d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260922
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36773}
The variable instance_count might be accessed from multiple threads when
different PeerConnectionFactory objects are used, which may create
multiple network threads. This is a pattern mostly noticed in tests.
This fixes issues with logging when run under TSAN, it should not have
any production impact.
Bug: chromium:1243702
Change-Id: Iab1412a7907545811a309cab27a3ae23b4718606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251983
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@google.com>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36045}
To disable dcSCTP and fallback to usrsctp, you can use the field trial
WebRTC-DataChannel-Dcsctp/Disabled/
Also remove a hidden no-break space in dcSCTP logging causing issues in
some log parsing.
Bug: chromium:1243702
Change-Id: I46136a8913a6d803a3c63c710f3ed29523e4d773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251867
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36027}
To disable dcSCTP and fallback to usrsctp, you can use the field trial
WebRTC-DataChannel-Dcsctp/Disabled/
Bug: chromium:1243702
Change-Id: Ia90b796562245558a61481317bcded437400b045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251800
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36018}
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 emphasizes the "hint" to potential external users that the
class has been deprecated.
Bug: webrtc:12339
Change-Id: Iab83481af69a505059297cce959f02b5ab649f2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237805
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35368}
Benchmarks will spam a lot of annoying lines because of this.
Bug: webrtc:13288
Change-Id: I1321abafb91baefcaa626a561bee58ca3f321291
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235371
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35225}
When sending a large amount of data, the sender will want to keep the
send buffer full so that the socket can quickly drain it as its able to
put more bytes on the wire.
When trying to send a message and when the send buffer is full, an error
will be returned and the OnError callback will be triggered. In these
situations, this is an expected and handled error and should not be
logged as an error, as it causes confusion.
Bug: chromium:1258225
Change-Id: I3e1feab03f60ba5c41cc524d8d8f066445030d18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235201
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35204}
Prior to this commit, the SCTP association could terminate due to too
many retransmission attempts when there is a long duration of packet
loss. The RTCPeerConnection wouldn't terminate, and when the network
later recovers (possibly using a different ICE candidate), it would be a
RTCPeerConnection with media, but without DataChannels.
This commit will make the dcSCTP library never abort by itself when
there are too many retransmissions. It will also put a cap on the retry
duration so that it will do a retry every three seconds (or lower).
Bug: webrtc:13129
Change-Id: I08162ea20d6a60aa0eae2717966d9a2ddba8fc22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232540
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35061}
In benchmarks, each log statement represent 2% of the CPU usage. RTC_LOG
is not very expensive, but not free either, and it's called for every
received and sent packet.
Bug: webrtc:12943
Change-Id: Id65baafb5e494091a3a7604687718fdd4f477d86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231223
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34929}
Using usrsctp_getladdrs() would sometimes be flagged by TSAN for a lock
order inversion. It was used to retrieve the "id" of the socket on the
transport.
The "id" is instead stored in the "ulp_info" parameter, which is
passed with each callback from usrsctp.
Bug: webrtc:12823
Change-Id: Ifb3d7780273a460e677526dd3a93f9365b29300c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229000
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34784}
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}
It is useful for more than just the transport.
Bug: webrtc:12961
Change-Id: Iad064c8fb707ca589a1c232e17436338fb06623d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225543
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34451}
When the transport is terminated, if an error has occured, it will
be propagated to the channels.
When such errors can happen at the SCTP level (e.g. out of resources),
RTCError may contain an error code matching the definition at
https://www.iana.org/assignments/sctp-parameters/sctp-parameters.xhtml#sctp-parameters-24
If the m= line is rejected or removed from SDP, an error will again be sent
to the data channels, signaling their unexpected transition to closed.
Bug: webrtc:12904
Change-Id: Iea3d8aba0a57bbedb5d03f0fb6f7aba292e92fe8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223541
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34386}
The factory allows us to isolate the implementation from users who only
need to depend directly on the public folder now.
Bug: webrtc:12614
Change-Id: Ied09cf772ed427eaf17a7b5705f587da57405640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220939
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34330}
for consistency with usrsctp_dumppacket which prefixes its output with a newline.
This makes the packets easier to grep and process with text2pcap.
BUG=webrtc:12614
Change-Id: I67bc2e0026250b21b030daf967ebc697640f2d7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220102
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34114}
Reduced the level so that the library can be run with INFO level without
a lot of spam. VERBOSE is still reserved for frequent logs.
Also, using WARNING for logs that are not fatal and which can easily
be triggered by the user.
Bug: webrtc:12614
Change-Id: If09c302b2b5bfc002471f86a8aeb74ba1172c705
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219465
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34054}
When verbose logs are enabled, SCTP packets will be dumped to debug
logs, allowing text2pcap to be used to generate PCAP files.
First start Chrome with verbose logs, and write those to file:
/path/to/chrome --enable-logging=stderr --v=4 2> out.log
Then extract the SCTP_PACKET traces and run text2pcap:
grep SCTP_PACKET out.log > sctp.log
text2pcap -n -i 132 -D -t '%H:%M:%S.' sctp.log sctp.pcapng
You may have to cut away more from the beginning if the debug logs
contain additional timestamps and more, e.g. like:
grep SCTP_PACKET out.log | cut -d ' ' -f 2- > sctp.log
Note that if there are multiple RTCPeerConnection objects created, each
will print out their packets to log, so to filter for a specific one:
grep "SCTP_PACKET DcSctpTransport0" out.log > sctp.log
Bug: webrtc:12614
Change-Id: Ibbceaf33719d09e7606247cb0496ddd827ea58bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218200
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33999}
This log statement is quite spammy and isn't of any help, so remove it.
Bug: webrtc:12614
Change-Id: Ia087228e0a3f602d558fcb7cbd9ec5295f35dcb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218202
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33981}
cricket::SendDataParams is replaced by webrtc::SendDataParams.
cricket::DataMessageType is replaced by webrtc::DataMessageType.
The sid member from cricket::SendDataParams is now passed as an argument
to functions that used one when necessary.
Bug: webrtc:7484
Change-Id: Ia4a89c9651fb54ab9a084a6098d49130b6319e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217761
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33966}
It doesn't make sense to use negative values or 0 to disable the
feature, so we use an optional int value.
Values bigger than 65535 are clamped down.
Bug: webrtc:12730
Change-Id: I6bd9cd92f7d0a70a78cf5a7c91dca52c28d08ba1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33954}
Add new PPIDs 56 and 57. When sending an empty message,
we use the corresponding PPID with a single byte data chunk.
On the receiving side, when detecting such a PPID, we just
ignore the payload content.
Bug: webrtc:12697
Change-Id: I6af481e7281db10d9663e1c0aaf97b3e608432a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215931
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33808}
This field was only used in RTP Data Channels and isn't needed anymore.
Bug: webrtc:6625
Change-Id: Ieaa7ae03ca3e90eb4ddec4d384f5a76cef1600cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215687
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33791}
