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webrtc / src / / 20e8cfb3414147f67047d2a0cbdb3aff6842b95c
commit20e8cfb3414147f67047d2a0cbdb3aff6842b95c[log] [tgz]
authorTaylor Brandstetter <deadbeef@webrtc.org>Thu May 24 23:43:02 2018
committerCommit Bot <commit-bot@chromium.org>Thu May 31 18:54:58 2018
tree4f8d3674594613e9da9f447cf297fa03c6abc29d
parent311428fecbc8b7e717391c60a71f43fe80645c47 [diff]
Fixing bug with PseudoTcp that corrupts data if initial packet is lost.

The issue occurred if a control segment is received after a non-control
segment received out-of-order, which only happens if:
* The initial "connect" segment is lost, and retransmitted later.
* Both sides send "connect"s simultaneously (rather than having
  designated server/client roles), such that the local side thinks a
  connection is established even before its "connect" has been
  acknowledged.
* Nagle algorithm disabled, allowing a data segment to be sent before
  the "connect" has been acknowledged.

This may seem like a pretty specific set of circumstances, but it can
happen with chromoting.

See the linked bug for more details.

Bug: webrtc:9208
Change-Id: I3cfe26e02158fcc5843f32d4e2ef7c511d58d9c9
Reviewed-on: https://webrtc-review.googlesource.com/78861
Reviewed-by: Sergey Ulanov <sergeyu@google.com>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23477}
2 files changed
tree: 4f8d3674594613e9da9f447cf297fa03c6abc29d
  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