Google Git
Sign in
webrtc / src / / 7c5541557b4e29dc6d203605b3c9b4b0362a1a4d
commit7c5541557b4e29dc6d203605b3c9b4b0362a1a4d[log] [tgz]
authorArtem Titov <titovartem@google.com>Thu Feb 28 09:25:52 2019
committerCommit Bot <commit-bot@chromium.org>Thu Feb 28 10:57:32 2019
tree96bc47dca1cb62bb43f0ea73f80db0773f02de78
parent03257b049e56358bf0b929d58fe59f57beef1ddf [diff]
Fix call setup: change way of adding media to the call.

To correctly send media from Bob to Alice, when Alice is calling Bob
we have to add transceivers for Bob's media to Alice first, because
it is forbidden to add new media sections into answer in Unified Plan,
so Alice's offer have to contain media sections for Bob's media tracks.

Bug: webrtc:10138
Change-Id: I8a5f19aa5ed6051a981472d5c79493362365f587
Reviewed-on: https://webrtc-review.googlesource.com/c/124492
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26892}
3 files changed
tree: 96bc47dca1cb62bb43f0ea73f80db0773f02de78
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. logging/
  10. media/
  11. modules/
  12. p2p/
  13. pc/
  14. resources/
  15. rtc_base/
  16. rtc_tools/
  17. sdk/
  18. stats/
  19. style-guide/
  20. system_wrappers/
  21. test/
  22. tools_webrtc/
  23. video/
  24. .clang-format
  25. .git-blame-ignore-revs
  26. .gitignore
  27. .gn
  28. .vpython
  29. abseil-in-webrtc.md
  30. AUTHORS
  31. BUILD.gn
  32. CODE_OF_CONDUCT.md
  33. codereview.settings
  34. common_types.h
  35. DEPS
  36. ENG_REVIEW_OWNERS
  37. LICENSE
  38. license_template.txt
  39. native-api.md
  40. OWNERS
  41. PATENTS
  42. PRESUBMIT.py
  43. presubmit_test.py
  44. presubmit_test_mocks.py
  45. pylintrc
  46. README.chromium
  47. README.md
  48. style-guide.md
  49. WATCHLISTS
  50. webrtc.gni
  51. 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