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webrtc / src.git / 29d16c0ed61e46b00b4becfd9e1d093d1551b11b
commit29d16c0ed61e46b00b4becfd9e1d093d1551b11b[log] [tgz]
authorPhilipp Hancke <philipp.hancke@googlemail.com>Thu Feb 03 11:24:52 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Thu Feb 03 12:18:50 2022
tree245725f38d6b74c7fb52caaff31eece111febf58
parent7b370b935ec0dac991da08f9da227df9ce245fd5 [diff]
stun/turn: use hostname when reconstructing the url

in the case of an ip address the hostname() call will return
that. This may also avoid leaking IP addresses from DNS resolutions
and is more similar to the url originally passed into the
peerconnection (but will for example produce a fully formed url and
resolves the port if none was given).

BUG=webrtc:13652

Change-Id: I000c66f7988b4b205e38c4dde5b888e48d8f6a0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250202
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35898}
4 files changed
tree: 245725f38d6b74c7fb52caaff31eece111febf58
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .style.yapf
  31. .vpython
  32. .vpython3
  33. AUTHORS
  34. BUILD.gn
  35. CODE_OF_CONDUCT.md
  36. codereview.settings
  37. DEPS
  38. DIR_METADATA
  39. ENG_REVIEW_OWNERS
  40. g3doc.lua
  41. LICENSE
  42. license_template.txt
  43. native-api.md
  44. OWNERS
  45. PATENTS
  46. PRESUBMIT.py
  47. presubmit_test.py
  48. presubmit_test_mocks.py
  49. pylintrc
  50. README.chromium
  51. README.md
  52. WATCHLISTS
  53. webrtc.gni
  54. webrtc_lib_link_test.cc
  55. 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 here 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