commit | af1f8655b2cb69af382396ea642eb0a2bf04bb4d | [log] [tgz] |
---|---|---|
author | Benjamin Wright <benwright@webrtc.org> | Mon Apr 01 18:25:23 2019 +0000 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Apr 01 19:11:07 2019 +0000 |
tree | 2d10c663ca022d9a8a152740289c2fa44241cf43 | |
parent | ae4b62318dbad797c2ca0a4f71aa5e65a2445a4e [diff] |
Revert "Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC." This reverts commit 7276b974b78ea4f409d8738b1b6f1515f7a8968e. 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}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.