Completed the functionalities of SrtpTransport.

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}
13 files changed
tree: 29349e1752968281ad0282585b57a7fb8824057c
  1. build_overrides/
  2. data/
  3. infra/
  4. resources/
  5. tools_webrtc/
  6. webrtc/
  7. .clang-format
  8. .git-blame-ignore-revs
  9. .gitignore
  10. .gn
  11. AUTHORS
  12. BUILD.gn
  13. CODE_OF_CONDUCT.md
  14. codereview.settings
  15. DEPS
  16. LICENSE
  17. license_template.txt
  18. LICENSE_THIRD_PARTY
  19. OWNERS
  20. PATENTS
  21. PRESUBMIT.py
  22. pylintrc
  23. README.md
  24. WATCHLISTS
README.md

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