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}
The rename ensures we don't confuse this implementation with
the new one based on the new dcSCTP library.
Bug: webrtc:12614
No-Presubmit: True
Change-Id: Ida08659bbea9c98aba8247d4368799ff7dd18729
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214482
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33688}
This reverts commit 8a38b1cf681cd77f0d59a68fb45d8dedbd7d4cee.
Reason for reland: Problem was identified; has something to do with
the unique_ptr with the custom deleter.
Original change's description:
> Revert "Fix race between destroying SctpTransport and receiving notification on timer thread."
>
> This reverts commit a88fe7be146b9b85575504d4d5193c007f2e3de4.
>
> Reason for revert: Breaks downstream test, still investigating.
>
> Original change's description:
> > Fix race between destroying SctpTransport and receiving notification on timer thread.
> >
> > This gets rid of the SctpTransportMap::Retrieve method and forces
> > everything to go through PostToTransportThread, which behaves safely
> > with relation to the transport's destruction.
> >
> > Bug: webrtc:12467
> > Change-Id: Id4a723c2c985be2a368d2cc5c5e62deb04c509ab
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208800
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33364}
>
> TBR=nisse@webrtc.org
>
> Bug: webrtc:12467
> Change-Id: Ib5d815a2cbca4feb25f360bff7ed62c02d1910a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209820
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33386}
Bug: webrtc:12467
Change-Id: I5f9fcd6df7a211e6edfa64577fc953833f4d9b79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210040
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33427}
This reverts commit a88fe7be146b9b85575504d4d5193c007f2e3de4.
Reason for revert: Breaks downstream test, still investigating.
Original change's description:
> Fix race between destroying SctpTransport and receiving notification on timer thread.
>
> This gets rid of the SctpTransportMap::Retrieve method and forces
> everything to go through PostToTransportThread, which behaves safely
> with relation to the transport's destruction.
>
> Bug: webrtc:12467
> Change-Id: Id4a723c2c985be2a368d2cc5c5e62deb04c509ab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208800
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33364}
TBR=nisse@webrtc.org
Bug: webrtc:12467
Change-Id: Ib5d815a2cbca4feb25f360bff7ed62c02d1910a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209820
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33386}
This gets rid of the SctpTransportMap::Retrieve method and forces
everything to go through PostToTransportThread, which behaves safely
with relation to the transport's destruction.
Bug: webrtc:12467
Change-Id: Id4a723c2c985be2a368d2cc5c5e62deb04c509ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33364}
The race occurs if the transport is being destroyed at the same time as
a callback occurs on the usrsctp timer thread (for example, for a
retransmission). Fixed by slightly extending the scope of mutex
acquisition to include posting a task to the network thread, where it's
safe to do further work.
Bug: chromium:1162424
Change-Id: Ia25c96fa51cd4ba2d8690ba03de8af9e9f1605ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33048}
When deinitializing usrsctp, we attempt to call usrsctp_finish in a loop
for three seconds (it may fail because another sctp thread is holding a
reference to something). If the three seconds run out, usrsctp is left
in a still initialized state, and bad things happen down the road if
usrsctp_init is called in the state.
Bug: chromium:1138878
Change-Id: I9c24d51d5a274b06bdf4183261694fc2989136c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32467}
When processing an INIT for existing SCTP association, which may happen
when transferring to a new remote endpoint, if there are chunks still on
the output queue, an SCTP_SEND_FAILED_EVENT notification will be
delivered while holding the INP lock. When processing this notification,
WebRTC was calling usrsctp_getladdrs in order to look up the
SctpTransport* associated with the socket, but this also tries to
acquire the INP lock, resulting in a deadlock.
The temporary fix is to simply not subscribe to the
SCTP_SEND_FAILED_EVENT notification; it was only used for logging
anyway.
Bug: chromium:1137936
Change-Id: I077efb4c769d6f7855d2fee2d266f19396b16f9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188502
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32427}
OnSctpInboundPacket is called not only for incoming packets, but for
notifications, which can be delivered on the usrsctp timer thread.
I suspect that these notifications can be delivered after we attempt to
close the socket, because if we attempt to close it while the timer
thread holds a reference, it isn't actually destroyed until the timer
thread finishes its operation.
Bug: chromium:1127774
Change-Id: Id6a883b14796e8f5bf1c2990f3d9d389d72c8a46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32204}
The signature of send_cb was changed, adding ulp_info. This change makes
it easier to retrieve the SctpTransport pointer from the callback.
Bug: webrtc:11899
Change-Id: I12a4ccd2d0deb329f6be17a4c7208449833dc188
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182984
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32067}
This is to allow testing without using the singleton sctp library.
cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.
patch 1 contain the original cl.
patch 2 modifications
Bug: none
Change-Id: Ic088da3eb7d9aada79e6d601dbf2d1aa2be777f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182840
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32024}
This reverts commit 4c0a381137c04fd80830af8a041e25e3428dd33f.
Reason for revert: Breaks downstream test
Original change's description:
> Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps
>
> This is to allow testing without using the singleton sctp library.
> cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
> Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.
>
> Bug: none
> Change-Id: I482241269463595062548870750d33f31238c6b1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182082
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32007}
TBR=deadbeef@webrtc.org,mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I46d5ba89fe723caccd065b0ac41d77ed45373838
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32008}
This is to allow testing without using the singleton sctp library.
cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.
Bug: none
Change-Id: I482241269463595062548870750d33f31238c6b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182082
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32007}
