e683c6871fef24d3ff64f085d6bc0e965f17fcf7
The SrtpTransport takes the SRTP responsibilities from the BaseChannel and SrtpFilter. SrtpTransport is now responsible for setting the crypto keys, protecting and unprotecting the packets. SrtpTransport doesn't know if the keys are from SDES or DTLS handshake. BaseChannel is now only responsible setting the offer/answer for SDES or extracting the key from DtlsTransport and configuring the SrtpTransport. SrtpFilter is used by BaseChannel as a helper for SDES negotiation. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2997983002 Cr-Commit-Position: refs/heads/master@{#19636}
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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
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