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webrtc / src / / f97afd65b25b67361284ad29345d685890ed9d1f
commitf97afd65b25b67361284ad29345d685890ed9d1f[log] [tgz]
authorYura Yaroshevich <yura.yaroshevich@gmail.com>Mon Apr 12 12:56:08 2021
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Apr 28 10:50:21 2021
treef948f9de4d42dd5c51dd166bd96ef431f9c25dff
parent2a605b13339db11a2e9052c35c41f899a53b098a [diff]
Reland "Add addr in error msg if stun sock sent with error"

This is a reland of 9ff75a6206142fbe17997bb092f8ec5485c2a084

Original change's description:
> Add addr in error msg if stun sock sent with error
>
> Before:
> ```
> (stun_port.cc:596): sendto : [0x00000041] No route to host
> ```
>
> After:
> ```
> (stun_port.cc:598): UDP send of 20 bytes to host stun1.l.google.com:19302 (74.125.200.127:19302) failed with error 65 : [0x00000041] No route to host
> ```
>
> Bug: None
> Change-Id: Ibcd487e97b37677225814562df30af66f655cddb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215000
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
> Cr-Commit-Position: refs/heads/master@{#33694}

Bug: None
Change-Id: Ic73327b20fd08ab41f378961333dd27f0f884525
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215926
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33862}
3 files changed
tree: f948f9de4d42dd5c51dd166bd96ef431f9c25dff
  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. style-guide/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .vpython
  32. abseil-in-webrtc.md
  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. style-guide.md
  53. WATCHLISTS
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. 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