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webrtc / src.git / cf6e24a12d30c6a64d8925ad03218f6bc1014343
commitcf6e24a12d30c6a64d8925ad03218f6bc1014343[log] [tgz]
authorZhi Huang <zhihuang@webrtc.org>Wed Feb 21 18:40:07 2018
committerCommit Bot <commit-bot@chromium.org>Wed Feb 21 19:18:19 2018
tree7371714c98372a5d0f3145bf89fe6226e9ce4fe2
parent52d86774c2d3c8d1b884c6c8edbe10e67483acff [diff]
Forward the SignalNetworkRouteChanged from DtlsSrtpTransport to BaseChannel.

In current implementation, the DtlsSrtpTransport listens to the
SignalNetworkRouteChanged but doesn't forward it to the BaseChannel which
makes it impossible for the media engine to update the network route and
the transport overhead.

The BaseChannel unit tests failed to catch this issue because it used a plain
unencrypted RTP transport for testing.

This CL fix that issue and update the BaseChannel tests.

Bug: webrtc:7013, b/73645191
Change-Id: I417b58ff9af4e3c4fac442ff10b5a85bc2093530
Reviewed-on: https://webrtc-review.googlesource.com/55940
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22140}
4 files changed
tree: 7371714c98372a5d0f3145bf89fe6226e9ce4fe2
  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. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. AUTHORS
  31. BUILD.gn
  32. CODE_OF_CONDUCT.md
  33. codereview.settings
  34. common_types.cc
  35. common_types.h
  36. DEPS
  37. LICENSE
  38. license_template.txt
  39. LICENSE_THIRD_PARTY
  40. native-api.md
  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.

Authoritative list of directories that contain the native API header files.

More info