commit | be66d95ab7f9428028806bbf66cb83800bda9241 | [log] [tgz] |
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author | Philipp Hancke <philipp.hancke@googlemail.com> | Mon Mar 01 13:56:22 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Mar 08 08:50:09 2021 |
tree | d4195386e2c77e3ebbdfa4342229599891ed820a | |
parent | 456a2642d36de9e35242ec50769582252e23b0ae [diff] |
srtp: document rationale for srtp overhead calculation documents why it is safe to not follow libsrtp's advice to ensure additional SRTP_MAX_TRAILER_LEN bytes are available when calling srtp_protect (and similar srtcp functions). BUG=None Change-Id: I504645d21553160f06133fd8bb3ee79e178247da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209064 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#33396}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.