Revert "Implement true negotiation for DatagramTransport with fallback to RTP."

This reverts commit 71c6482baf0ff17141c635e6a7639493db68a65c.

Reason for revert: Lands too much at once and breaks downstream tests that need to implement new interfaces first.

Original change's description:
> Implement true negotiation for DatagramTransport with fallback to RTP.
> 
> In short, the caller places a x-opaque line in SDP for each m= section that
> uses datagram transport.  If the answerer supports datagram transport, it will
> parse this line and create a datagram transport.  It will then echo the x-opaque
> line into the answer (to indicate that it accepted use of datagram transport).
> 
> If the offer and answer contain exactly the same x-opaque line, both peers will
> use datagram transport.  If the x-opaque line is omitted from the answer (or is
> different in the answer) they will fall back to RTP.
> 
> Note that a different x-opaque line in the answer means the answerer did not
> understand something in the negotiation proto.  Since WebRTC cannot know what
> was misunderstood, or whether it's still possible to use the datagram transport,
> it must fall back to RTP.  This may change in the future, possibly by passing
> the answer to the datagram transport, but it's good enough for now.
> 
> Negotiation consists of four parts:
>  1. DatagramTransport exposes transport parameters for both client and server
>  perspectives.  The client just echoes what it received from the server (modulo
>  any fields it might not have understood).
> 
>  2. SDP adds a x-opaque line for opaque transport parameters.  Identical to
>  x-mt, but this is specific to datagram transport and goes in each m= section,
>  and appears in the answer as well as the offer.
>   - This is propagated to Jsep as part of the TransportDescription.
>   - SDP files: transport_description.h,cc, transport_description_factory.h,cc,
>     media_session.cc, webrtc_sdp.cc
> 
>  3. JsepTransport/Controller:
>   - Exposes opaque parameters for each mid (m= section).  On offerer, this means
>     pre-allocating a datagram transport and getting its parameters.  On the
>     answerer, this means echoing the offerer's parameters.
>   - Uses a composite RTP transport to receive from either default RTP or
>     datagram transport until both offer and answer arrive.
>   - If a provisional answer arrives, sets the composite to send on the
>     provisionally selected transport.
>   - Once both offer and answer are set, deletes the unneeded transports and
>     keeps whichever transport is selected.
> 
>  4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
> 
> Bug: webrtc:9719
> Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28182}

TBR=steveanton@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org

Change-Id: I0d502c4a6d27516c35ed85154f3fa5869f88b3b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140822
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28188}
23 files changed
tree: 3e18d103002f1136c11482270c0a05bcdd1d4164
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. crypto/
  8. data/
  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. 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