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webrtc / src / / 2d5f3cb93336b0fcd2e3efff2671b2a034821f64
commit2d5f3cb93336b0fcd2e3efff2671b2a034821f64[log] [tgz]
authorBenjamin Wright <benwright@webrtc.org>Tue May 22 21:46:06 2018
committerCommit Bot <commit-bot@chromium.org>Wed May 23 00:28:39 2018
tree67ef4a571c0060a1f865c64ccdd4a21eea385d28
parent02c65869c3bae25567eb2681ec78bb1f6d2322c9 [diff]
Added an integration test to validate TURN servers can send media in relay mode.

End to end test for media sent over a TCP TURN server with both clients in relay
This test validates that media can be sent between two clients who are set up
to relay information with the server configured to use TCP instead of UDP.

Bug: webrtc:7668
Change-Id: I3efd04048589c144494f90f2cdf3df5f9f80300e
Reviewed-on: https://webrtc-review.googlesource.com/76507
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23354}
1 file changed
tree: 67ef4a571c0060a1f865c64ccdd4a21eea385d28
  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. style-guide/
  22. system_wrappers/
  23. test/
  24. third_party/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .vpython
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  36. common_types.cc
  37. common_types.h
  38. DEPS
  39. LICENSE
  40. license_template.txt
  41. LICENSE_THIRD_PARTY
  42. native-api.md
  43. OWNERS
  44. PATENTS
  45. PRESUBMIT.py
  46. presubmit_test.py
  47. presubmit_test_mocks.py
  48. pylintrc
  49. README.chromium
  50. README.md
  51. style-guide.md
  52. THIRD_PARTY_CHROMIUM_DEPS.json
  53. THIRD_PARTY_WEBRTC_DEPS.json
  54. typedefs.h
  55. WATCHLISTS
  56. webrtc.gni
  57. 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