Changes to enable use of DatagramTransport as a data channel transport.

PeerConnection now has a new setting in RTCConfiguration to enable use of
datagram transport for data channels.  There is also a corresponding field
trial, which has both a kill-switch and a way to change the default value.

PeerConnection's interaction with MediaTransport for data channels has been
refactored to work with DataChannelTransportInterface instead.

Adds a DataChannelState and OnStateChanged() to the DataChannelSink
callbacks.  This allows PeerConnection to listen to the data channel's
state directly, instead of indirectly by monitoring media transport
state.  This is necessary to enable use of non-media-transport (eg.
datagram transport) data channel transports.

For now, PeerConnection watches the state through MediaTransport as well.
This will persist until MediaTransport implements the new callback.

Datagram transport use is negotiated.  As such, an offer that requests to use
datagram transport for data channels may be rejected by the answerer.  If the
offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data
channels with an extra x-opaque parameter for datagram transport.  If the
opaque parameter is rejected (by an answerer without datagram support), the
offerer may fall back to SCTP.

If DTLS is not enabled, there is no viable fallback.  In this case, the data
channels are negotiated as media transport data channels.  If the receiver does
not understand the x-opaque line, it will reject these data channels, and the
offerer's data channels will be closed.

Bug: webrtc:9719
Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28932}
20 files changed
tree: f48f5a6970f0d8405196efc728fb727289ab9c98
  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