cd3fc5d90cb3914a4329b32a3d319824d5f5eddf
The DtlsSrtpTransport takes the reponsiblity of setting up DTLS-SRTP from the BaseChannel. The BaseChannel doesn't handle the signals from the P2P layer transport anymore. The RtpTransport handles the signals from the PacketTransportInternal and the DtlsSrtpTransport handles the DTLS-specific signals and determines when to extract the keys and setting the parameters. In channel_unittests.cc, call from DTLS to SDES is expected to fail since the fallback from DTLS to SDES is not supported. Bug: webrtc:7013 Change-Id: I0a54e017986f5a8ae9710e79643a4651bef3c38f Reviewed-on: https://webrtc-review.googlesource.com/24702 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20941}
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
Description
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