and use uint8_t instead of unsigned char. Follow-up from
https://webrtc-review.googlesource.com/c/src/+/365274
BUG=webrtc:357776213
Change-Id: Ibc97e5cc85316ba69b4133b7f3c42e3afbdd7abd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365540
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43263}
This is a reland of commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d
with more backward compat which also fixes the off-by-one issue which caused wrong SRTP keys to be extracted.
Original change's description:
> Spanify SRTP key export
>
> and simplify the interface used as this is only used for exporting
> SRTP keys and passing arcane OpenSSL arguments around does not make
> much sense.
>
> BUG=webrtc:357776213
>
> Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43198}
Bug: webrtc:357776213
Change-Id: I5d43dc23f90ef630834fb400751979fcc5e18203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43225}
This reverts commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d.
Reason for revert: breaks downstream compilation
Original change's description:
> Spanify SRTP key export
>
> and simplify the interface used as this is only used for exporting
> SRTP keys and passing arcane OpenSSL arguments around does not make
> much sense.
>
> BUG=webrtc:357776213
>
> Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43198}
Bug: webrtc:357776213
Change-Id: I03ffcda3d6821718f355b243ce78a9c54b4036f3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365062
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43202}
and simplify the interface used as this is only used for exporting
SRTP keys and passing arcane OpenSSL arguments around does not make
much sense.
BUG=webrtc:357776213
Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43198}
since it contains helpers mostly related to cryptographically secure random numbers and strings.
BUG=webrtc:339300437
Change-Id: I10db939534b25dc792ac1600a4721d1b84521880
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42441}
since the tie breaker is owned by the allocator now.
BUG=webrtc:42224914
Change-Id: I76bd5ae714fb2a6df38e014991242f390ae87e6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42371}
This is a reland of commit 02b5f3c9c12cddf3fc6e9125238b77ddb44f3b53
without making SetRemoteFingerprint private (but adding a deprecation warning)
Original change's description:
> dtls: allow dtls role to change during DTLS restart
>
> which is characterized by a change in remote fingerprint and
> causes a new DTLS handshake. This allows renegotiating the
> client/server role as well.
> Spec guidance is provided by
> https://www.rfc-editor.org/rfc/rfc5763#section-6.6
>
> BUG=webrtc:5768
>
> Change-Id: I0e8630c0c5907cc92720762a4320ad21a6190d28
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271680
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#37821}
Bug: webrtc:5768
Change-Id: I8dd674db8b683160013e1b4aa7776775d130978f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272221
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#37838}
This reverts commit 02b5f3c9c12cddf3fc6e9125238b77ddb44f3b53.
Reason for revert: SetRemoteFingerprint called by downstream code.
Original change's description:
> dtls: allow dtls role to change during DTLS restart
>
> which is characterized by a change in remote fingerprint and
> causes a new DTLS handshake. This allows renegotiating the
> client/server role as well.
> Spec guidance is provided by
> https://www.rfc-editor.org/rfc/rfc5763#section-6.6
>
> BUG=webrtc:5768
>
> Change-Id: I0e8630c0c5907cc92720762a4320ad21a6190d28
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271680
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#37821}
Bug: webrtc:5768
Change-Id: I266b7fdc9cc0b6dc9d3fa732fca37407b98e0816
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272220
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>
Cr-Commit-Position: refs/heads/main@{#37822}
which is characterized by a change in remote fingerprint and
causes a new DTLS handshake. This allows renegotiating the
client/server role as well.
Spec guidance is provided by
https://www.rfc-editor.org/rfc/rfc5763#section-6.6
BUG=webrtc:5768
Change-Id: I0e8630c0c5907cc92720762a4320ad21a6190d28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#37821}
There is a suspicion that it causes OpenSSL errors:
[openssl_stream_adapter.cc(961)]
OpenSSLStreamAdapter::Error(SSL_write, 5, 0)
This commit does change the interaction with OpenSSL as propagating the
socket write errors as SR_BLOCK results in calling BIO_set_retry_write,
as part of current implementation of OpenSSLStreamAdapter.
Testing this regression has proven to be hard to do manually.
This reverts commit edfaaef086ccff2dbff29d64c9a8d9f633637c57.
Bug: webrtc:12943
Change-Id: Ib6767bd4af68c59fd3b7cb051341876f175bb921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230420
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34875}
The UDP sockets in WebRTC are non-blocking, and when writing too much
to them so that their send buffer becomes exhausted, they will return
EAGAIN or EWOULDBLOCK, which indicates that the client will need to
retry a bit later.
Media packets are generally sent by the pacer, which generally avoids
this exhaustion, but for SCTP which has a congestion control algorithm
quite similar to TCP, it may overshoot the amount of data it writes. If
the SCTP library can be notified when writing fails, it can stop writing
for a while until the socket recovers, which will result in less
overshooting and fewer lost packets (possibly even none). But for the
SCTP library to be able to know this, errors must be propagated, which
they weren't with the argument that packets may get lost anyway.
Bug: webrtc:12943
Change-Id: I9244580abf0d48ff810da30a23f995d12be623ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228439
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34751}
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).
This CL renames some constants and follows the C++ style guide.
Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
Make a few more members const, remove members that aren't used,
set max ssl version number on construction and remove setter.
Bug: none
Change-Id: I6c1a7cabf1e795e027f1bc53b994517e9aef0e93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33622}
It does not make sense for DtlsTransport to own ICE, and this arrangement will
not work when negotiating datagram or DTLS transport. During negotiation, both
a DTLS transport and a datagram transport need to be ready to receive from the
same ICE transport, depending on which protocol is chosen by the answerer. Once
the answerer chooses a protocol, the transport that is not chosen must be
deleted, but ICE must be left intact for use by the remaining transport.
Bug: webrtc:9719
Change-Id: Ibab969b574c981e3834ced71f8ff88008cb26a6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139340
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28113}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
