This patchs adds a field trial for enabling DTLS1.3, WebRTC-ForceDtls13
- "Enabled" set max version to DTLS1.3
- "Only" set min & max version to DTLS1.3
Wireup a FieldTrialsView so that this does not
use the global string.
Also convert the WebRTC-DisableTlsSessionTicketKillswitch
from global string to FieldTrialsView.
BUG=webrtc:383141571
Change-Id: Ia775efc1dcbffd01bfddb6030490438cb8de89d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372261
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43616}
- add DTLS1.3 ciphers (without KeyType)
- remove code in dtls_transport.cc that tries to parse DTLS packet
- cleanup some test
- start on test for packet loss during dtls handshake (more to come!)
After this patch is submitted, it is possible
to set max version = dtls1.3 and it will active
but DON'T do it yet.
BUG=webrtc:383141571
Change-Id: I6f9a120c53415ccee7a560ea83bd0c2636702997
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371300
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43595}
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}
Move it away from the "proprietary" SSL_CIPHER_get_id and looking up the cipher based on that towards SSL_CIPHER_standard_name.
SSL_CIPHER_get_id and the associated GetSslCipherSuite API is kept around for
WebRTC.PeerConnection.SslCipherSuite.*
UMA metrics and metrics compability (despite not yielding the IANA ids it promises).
BUG=None
Change-Id: Iaa357e3e31dc90abea688cf6ca10c0b40582ef38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363202
Reviewed-by: David Benjamin <davidben@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43097}
Remove internal use of SignalSSLHandshakeError and prepare removal of
sigslot dependency from SSLStreamAdapter.
Bug: webrtc:11943
Change-Id: I9768e2e31529945620bdd8d0d285042bb2388b7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40695}
This is part of a large-scale effort to increase adoption of
absl::string_view across the WebRTC code base.
This CL converts the majority of "const std::string&"s in function
parameters under rtc_base/ to absl::string_view.
Bug: webrtc:13579
Change-Id: I2b1e3776aa42326aa405f76bb324a2d233b21dca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254081
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Anders Lilienthal <andersc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36239}
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}
