2ebbff83ee
Move the GCM srtp cipher suites below the default SRTP_AES128_CM_SHA1_80 one. This will not negotiate them by default since they have an impact on packet overhead for audio-only calls. GCM can still be negotiated if the peer offers it as preferred cipher suite or answers with just that cipher suite. BUG=chromium:713701 Change-Id: I79bd4ab827e5c7f55f5550d14db3f4217a7eff86 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158404 Reviewed-by: Justin Uberti <juberti@google.com> Reviewed-by: Justin Uberti <juberti@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Justin Uberti <juberti@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29672}
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