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webrtc / src / / 5150ee40f45fe9cdef03b4454bfde7e099860673
commit5150ee40f45fe9cdef03b4454bfde7e099860673[log] [tgz]
authorSeth Hampson <shampson@webrtc.org>Thu Apr 05 00:39:19 2018
committerCommit Bot <commit-bot@chromium.org>Thu Apr 05 20:08:05 2018
treed1d6b44bcbea794be42aba3c181ebeaa89753dc6
parent09a6cd55416e9ff8826d98c4de024b1cfa22104d [diff]
Changing MTU size for SCTP socket options.

With the latest usrsctp roll, the MTU value you provide is the space
avaiable for chunks in the packet. We previously specified this to be the
MTU for the entire SCTP packet, so we were logging errors when the SCTP
packets were 12 bytes larger than expected (the size of the SCTP header).
This fix updates our MTU specified to account for the SCTP header size
as well.

Bug: webrtc:9082
Change-Id: Id3bfa839d4e7662230111ebbdf33bd81ccdc7cf4
Reviewed-on: https://webrtc-review.googlesource.com/66943
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22754}
1 file changed
tree: d1d6b44bcbea794be42aba3c181ebeaa89753dc6
  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. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.cc
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. LICENSE_THIRD_PARTY
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. style-guide.md
  51. typedefs.h
  52. WATCHLISTS
  53. webrtc.gni
  54. 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