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webrtc / src / eb23e1779822ef34df5f62903e19741b76cafdb9
commiteb23e1779822ef34df5f62903e19741b76cafdb9[log] [tgz]
authorzhihuang <zhihuang@webrtc.org>Tue Sep 19 08:12:52 2017
committerCommit Bot <commit-bot@chromium.org>Tue Sep 19 08:12:52 2017
tree7e9882bc598856085ed1acd69d66e4cb8be0dc45
parent25724047892db5bc6b1f5bb0af46886152a46bda [diff]
Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ )

Reason for revert:
This seems to be causing some video freezes. See https://bugs.chromium.org/p/webrtc/issues/detail?id=8251

Original issue's description:
> 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}
> Committed: https://chromium.googlesource.com/external/webrtc/+/e683c6871fef24d3ff64f085d6bc0e965f17fcf7

TBR=deadbeef@webrtc.org,pthatcher@google.com,zhihuang@webrtc.org
Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7013

Review-Url: https://codereview.webrtc.org/3018513002
Cr-Commit-Position: refs/heads/master@{#19895}
13 files changed
tree: 7e9882bc598856085ed1acd69d66e4cb8be0dc45
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. voice_engine/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.cc
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. LICENSE_THIRD_PARTY
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. typedefs.h
  51. WATCHLISTS
  52. webrtc.gni
  53. whitespace.txt
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