Revert "Fix RTP header extension encryption"

This reverts commit a743303211b89bbcf4cea438ee797bbbc7b59e80.

Reason for revert: Breaks downstream tests that attempt to call FindHeaderExtensionByUri with 2 arguments. Could you keep the old 2-argument method declaration and just forward the call to the new 3-argument method with a suitable no-op filter?

Original change's description:
> Fix RTP header extension encryption
>
> Previously, RTP header extensions with encryption had been filtered
> if the encryption had been activated (not the other way around) which
> was likely an unintended logic inversion.
>
> In addition, it ensures that encrypted RTP header extensions are only
> negotiated if RTP header extension encryption is turned on. Formerly,
> which extensions had been negotiated depended on the order in which
> they were inserted, regardless of whether or not header encryption was
> actually enabled, leading to no extensions being sent on the wire.
>
> Further changes:
>
> - If RTP header encryption enabled, prefer encrypted extensions over
>   non-encrypted extensions
> - Add most extensions to list of extensions supported for encryption
> - Discard encrypted extensions in a session description in case encryption
>   is not supported for that extension
>
> Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
> into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
> header extensions will prevent any RTP packets being sent/received.
>
> Bug: webrtc:11713
> Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33723}

TBR=deadbeef@webrtc.org,terelius@webrtc.org,hta@webrtc.org,lennart.grahl@gmail.com

Change-Id: I7df6b0fa611c6496dccdfb09a65ff33ae4a52b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215222
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33727}
12 files changed
tree: f07ec7cc01ac52da161a1ae180e9b10158bfc850
  1. .clang-format
  2. .git-blame-ignore-revs
  3. .gitignore
  4. .gn
  5. .vpython
  6. AUTHORS
  7. BUILD.gn
  8. CODE_OF_CONDUCT.md
  9. DEPS
  10. DIR_METADATA
  11. ENG_REVIEW_OWNERS
  12. LICENSE
  13. OWNERS
  14. PATENTS
  15. PRESUBMIT.py
  16. README.chromium
  17. README.md
  18. WATCHLISTS
  19. abseil-in-webrtc.md
  20. api/
  21. audio/
  22. build_overrides/
  23. call/
  24. codereview.settings
  25. common_audio/
  26. common_video/
  27. data/
  28. docs/
  29. examples/
  30. g3doc.lua
  31. g3doc/
  32. license_template.txt
  33. logging/
  34. media/
  35. modules/
  36. native-api.md
  37. net/
  38. p2p/
  39. pc/
  40. presubmit_test.py
  41. presubmit_test_mocks.py
  42. pylintrc
  43. resources/
  44. rtc_base/
  45. rtc_tools/
  46. sdk/
  47. stats/
  48. style-guide.md
  49. style-guide/
  50. system_wrappers/
  51. test/
  52. tools_webrtc/
  53. video/
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. 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 here 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