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}
12 files changed
tree: 09ac2dce4b3532cf7a3e3a41a018c6298b5cbe74
  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