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webrtc / src / 219d8ce889d174f1cef84a3d8779ee3cbf4ca1aa
commit219d8ce889d174f1cef84a3d8779ee3cbf4ca1aa[log] [tgz]
authorJonas Oreland <jonaso@webrtc.org>Wed Jan 15 14:29:48 2020
committerCommit Bot <commit-bot@chromium.org>Wed Jan 15 16:09:38 2020
tree80cc8c5c44491cc415137d0ede54f88a63066784
parenta846cef1974807229e7b22b4eaf5ee417e030baf [diff]
GOOG_PING: improve handshake

This patch improves handshake wrt GOOG_PING support so that
- if goog_ping_enable: sender send it's goog-ping version until it gets
STUN_BINDING_RESPONSE
- receiver only sends it's goog-ping-version if getting a
goog-ping-version in the request

This means that the overhead of STUN_ATTR_GOOG_MISC_INFO is only
- added on STUN_BINDING_REQUEST until a response is received.
- added on STUN_BINDING_RESPONSE if remote peer request it.

This is wire compatible with older versions so that
- new sender will enable GOOG_PING with new/old receiver.
- old sender will enable GOOG_PING with old receiver.
- old version will not enable GOOG_PING with new receiver
  (receiver expecting sender to announce first).

BUG: webrtc:11100
Change-Id: Ib3434c593988188150f4c7506918139aaf138d0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165787
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30269}
4 files changed
tree: 80cc8c5c44491cc415137d0ede54f88a63066784
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  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. webrtc_lib_link_test.cc
  53. 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