Google Git
Sign in
webrtc / src / / 4436887ed2d3324279e0f2e091c9e9355392721a
commit4436887ed2d3324279e0f2e091c9e9355392721a[log] [tgz]
authorGuido Urdaneta <guidou@webrtc.org>Fri May 31 09:32:01 2019
committerCommit Bot <commit-bot@chromium.org>Fri May 31 10:16:10 2019
treee5902a55e15a279859746cfad2a06705d7dc2538
parenta937c6ea21799b36866a58a8430aad8d032c0e1c [diff]
Revert "Change default secure SCTP protocol to UDP/DTLS/SCTP"

This reverts commit c3f4820e129d44471b366b8885a67b5392918d5a.

Reason for revert: Will temporarily revert to fix an issue and reland afterwards.

Original change's description:
> Change default secure SCTP protocol to UDP/DTLS/SCTP
> 
> The old value - DTLS/SCTP - is not standards conformant,
> and the new value should be parsable since Chrome M61.
> 
> Bug: webrtc:7706
> Change-Id: I7468cc9597dec4ef4b102fccddc4e981fed7e8d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136804
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27940}

TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:7706
Change-Id: Ida8ae20767485c75edc44dff8a3fa1af2006f207
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139244
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28121}
4 files changed
tree: e5902a55e15a279859746cfad2a06705d7dc2538
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. crypto/
  8. data/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  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. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  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. WATCHLISTS
  51. webrtc.gni
  52. 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