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}
Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
security gains, and will provide binary size improvements as well once
the default list of built-in certificates can be removed; the code
dealing with them still depends on the X509 API.
Implemented by splitting openssl_identity and openssl_certificate
into BoringSSL and vanilla OpenSSL implementations.
No-Try: True
Bug: webrtc:11410
Change-Id: I86ddb361b94ad85b15ebb8743490de83632ca53f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196941
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32818}
This reverts commit 72f638a9a2.
Reason for revert: downstream build failures
Original change's description:
> Use CRYPTO_BUFFER APIs instead of X509 when building with BoringSSL.
>
> Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
> security gains, and will provide binary size improvements as well once
> the default list of built-in certificates can be removed; the code
> dealing with them still depends on the X509 API.
>
> Implemented by splitting openssl_identity and openssl_certificate
> into BoringSSL and vanilla OpenSSL implementations.
>
> Bug: webrtc:11410
> Change-Id: Idc043462faac5e4ab1b75bedab2057197f80aba6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174120
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: David Benjamin <davidben@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32811}
TBR=deadbeef@webrtc.org,mbonadei@webrtc.org,davidben@webrtc.org,hta@webrtc.org
Change-Id: Ib5e55cb5798a2f3d25a4460f5311d2e650d3fa82
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11410
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196742
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32812}
Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
security gains, and will provide binary size improvements as well once
the default list of built-in certificates can be removed; the code
dealing with them still depends on the X509 API.
Implemented by splitting openssl_identity and openssl_certificate
into BoringSSL and vanilla OpenSSL implementations.
Bug: webrtc:11410
Change-Id: Idc043462faac5e4ab1b75bedab2057197f80aba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174120
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32811}
This includes relying on related types such as MessageData and
PostEvent functionality inside the StreamInterface itself.
This affects mostly tests but OpenSSLStreamAdapter
requires special attention.
Bug: webrtc:11988
Change-Id: Ib5c895f1bdf77bb49e3162bd49718f8a98812d91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185505
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32290}
This reverts commit af1f8655b2
Landing the change with default set to
"enabled" (DTLS 1.0 will continue to work by default),
so that flipping the default can be a separate CL.
Original change's description:
> Revert "Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC."
>
> This reverts commit 7276b974b7.
>
> Reason for revert: Changing to a later Chrome release.
>
> Original change's description:
> > Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
> >
> > This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
> > is part of a larger effort at Google to remove old TLS protocols:
> > https://security.googleblog.com/2018/10/modernizing-transport-security.html
> >
> > For the M74 timeline I have added a disabled by default field trial
> > WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
> > as consumers move away from these legacy cipher protocols but it will be off
> > in Chrome.
> >
> > This is compliant with the webrtc-security-arch specification which states:
> >
> > All Implementations MUST implement DTLS 1.2 with the
> > TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
> > curve [FIPS186]. Earlier drafts of this specification required DTLS
> > 1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
> > at the time of this writing some implementations do not support DTLS
> > 1.2; endpoints which support only DTLS 1.2 might encounter
> > interoperability issues. The DTLS-SRTP protection profile
> > SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
> > Implementations MUST favor cipher suites which support (Perfect
> > Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
> > over non-AEAD cipher suites.
> >
> > Bug: webrtc:10261
> > Change-Id: I847c567592911cc437f095376ad67585b4355fc0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
> > Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> > Reviewed-by: David Benjamin <davidben@webrtc.org>
> > Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27006}
>
> TBR=steveanton@webrtc.org,davidben@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10261
> Change-Id: I34727e65c069e1fb2ad71838828ad0a22b5fe811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130367
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27403}
Bug: webrtc:10261
Change-Id: I28c6819d37665976e396df280b4abf48fb91d533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169851
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30733}
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.
Bug: webrtc:9883
Change-Id: I3cb857cc707d5e897759366d1478cc1ec19bce9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165344
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30180}
Bug: chromium:1018077
Change-Id: I585d4064f39e5f9d268b408ebf6ae13a056c778a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158403
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29628}
This reverts commit 7276b974b7.
Reason for revert: Changing to a later Chrome release.
Original change's description:
> Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
>
> This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
> is part of a larger effort at Google to remove old TLS protocols:
> https://security.googleblog.com/2018/10/modernizing-transport-security.html
>
> For the M74 timeline I have added a disabled by default field trial
> WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
> as consumers move away from these legacy cipher protocols but it will be off
> in Chrome.
>
> This is compliant with the webrtc-security-arch specification which states:
>
> All Implementations MUST implement DTLS 1.2 with the
> TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
> curve [FIPS186]. Earlier drafts of this specification required DTLS
> 1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
> at the time of this writing some implementations do not support DTLS
> 1.2; endpoints which support only DTLS 1.2 might encounter
> interoperability issues. The DTLS-SRTP protection profile
> SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
> Implementations MUST favor cipher suites which support (Perfect
> Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
> over non-AEAD cipher suites.
>
> Bug: webrtc:10261
> Change-Id: I847c567592911cc437f095376ad67585b4355fc0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: David Benjamin <davidben@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27006}
TBR=steveanton@webrtc.org,davidben@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10261
Change-Id: I34727e65c069e1fb2ad71838828ad0a22b5fe811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130367
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27403}
This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
is part of a larger effort at Google to remove old TLS protocols:
https://security.googleblog.com/2018/10/modernizing-transport-security.html
For the M74 timeline I have added a disabled by default field trial
WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
as consumers move away from these legacy cipher protocols but it will be off
in Chrome.
This is compliant with the webrtc-security-arch specification which states:
All Implementations MUST implement DTLS 1.2 with the
TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
curve [FIPS186]. Earlier drafts of this specification required DTLS
1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
at the time of this writing some implementations do not support DTLS
1.2; endpoints which support only DTLS 1.2 might encounter
interoperability issues. The DTLS-SRTP protection profile
SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
Implementations MUST favor cipher suites which support (Perfect
Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
over non-AEAD cipher suites.
Bug: webrtc:10261
Change-Id: I847c567592911cc437f095376ad67585b4355fc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27006}
