Revert of If SRTP sessions exist, don't create new ones when applying answer. (patchset #1 id:1 of https://codereview.webrtc.org/3019443002/ )

Reason for revert:
Speculative revert since all Android bots on WebRTC FYI started to fail when this CL landed.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus6%29

Original issue's description:
> If SRTP sessions exist, don't create new ones when applying answer.
>
> Instead, call the "Update" methods of SrtpSession, which will just call
> srtp_update, instead of wiping out the session state completely.
>
> This was causing decryption to stop working when subsequent
> offers/answers are applied. We don't know enough about SRTP to
> understand the root cause, and I wasn't able to write an integration
> test that reproduces the issue... But at least this fixes the bug that
> can be reproduced reliably using Hangouts.
>
> BUG=webrtc:8251
>
> Review-Url: https://codereview.webrtc.org/3019443002
> Cr-Commit-Position: refs/heads/master@{#19874}
> Committed: https://webrtc.googlesource.com/src/+/5ada7acf603e90e71990e9d4ff8f49389f24958c

TBR=zhihuang@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8251
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/3017543002
Cr-Commit-Position: refs/heads/master@{#19882}
1 file changed
tree: c42ab6f696297de8c78be2b0d7ab3df6c1725662
  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