137991396d
This reverts commit af1f8655b2cb69af382396ea642eb0a2bf04bb4d 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 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} 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}
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct