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Florent Castelli a6983c6ea2 sctp: Add DcsctpTransport based on dcSCTP 
Bug: webrtc:12614
Change-Id: Ie710621610fff9f8bb6c7d800419675892d6a70c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215680
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33935}
2021-05-06 09:38:49 +00:00
api Refactor the PlatformThread API. 2021-05-05 09:59:07 +00:00
audio Use Timestamp to represent packet receive timestamps 2021-05-04 13:16:54 +00:00
build_overrides Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492) 2021-04-12 18:25:58 +00:00
call Update WebRTC code version (2021-05-06T04:04:14). 2021-05-06 05:53:52 +00:00
common_audio Avoid undefined behavior in a division operation. 2021-04-23 07:49:24 +00:00
common_video Reland "Remove Invoke from VideoChannel::FillBitrateInfo." 2021-05-03 15:16:34 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Create a VideoFrameTrackingId RTP header extension. 2021-03-25 17:25:18 +00:00
examples Revert "Ensure method which updates UI is called in main thread" 2021-04-30 09:26:03 +00:00
g3doc Add conceptual documentation for Audio - Mixer 2021-04-22 08:59:38 +00:00
logging Remove use of istream in RTC event log parser. 2021-03-31 13:21:58 +00:00
media sctp: Add DcsctpTransport based on dcSCTP 2021-05-06 09:38:49 +00:00
modules Audio - Mixer conceptual documentation: fix footnote 2021-05-06 09:17:19 +00:00
net/dcsctp dcsctp: Handle starting timer from timer callback 2021-05-05 13:13:03 +00:00
p2p Remove unused setter for Port::socket_factory() 2021-05-04 15:07:48 +00:00
pc sctp: Add DcsctpTransport based on dcSCTP 2021-05-06 09:38:49 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Refactor the PlatformThread API. 2021-05-05 09:59:07 +00:00
rtc_tools Replace more instances of rtc::RefCountedObject with make_ref_counted. 2021-04-27 17:01:59 +00:00
sdk Refactor the PlatformThread API. 2021-05-05 09:59:07 +00:00
stats Simplify reference counting implementation of PendingTaskSafetyFlag. 2021-04-21 07:04:01 +00:00
style-guide Remove kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00
system_wrappers Consolidate the different NTP clocks into one. 2021-04-08 13:54:04 +00:00
test Refactor the PlatformThread API. 2021-05-05 09:59:07 +00:00
tools_webrtc crc32c: Point the licensing script to the LICENSE file 2021-05-03 16:46:30 +00:00
video Store RtpPacketReceived::arrival_time as Timestamp. 2021-05-05 16:22:33 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .cache to .gitignore. 2021-01-20 15:01:07 +00:00
.gn Enable GN check on //net. 2021-05-03 14:23:09 +00:00
.vpython Update six library version 2021-04-26 16:39:07 +00:00
abseil-in-webrtc.md Polish the "Using Abseil in WebRTC" docs 2020-10-16 13:42:00 +00:00
AUTHORS Revert "Ensure method which updates UI is called in main thread" 2021-04-30 09:26:03 +00:00
BUILD.gn Move RTC_ENABLE_WIN_WGC define to the top level BUILD.gn 2021-04-08 16:31:49 +00:00
CODE_OF_CONDUCT.md
codereview.settings Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
DEPS Roll chromium_revision 293ea3a1c4..69e6dca23e (879621:879735) 2021-05-06 08:43:59 +00:00
DIR_METADATA Move metadata in OWNERS files to DIR_METADATA files. 2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS Remove kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00
g3doc.lua Improve webrtc documentation infra. Preview at: 2021-03-30 10:29:30 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
native-api.md Make the remote_bitrate_estimator build target private 2020-11-26 12:21:22 +00:00
OWNERS Add mbonadei@ as owner of .pylintrc / .vpython. 2021-04-26 16:34:57 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py sctp: Rename SctpTransport to UsrSctpTransport 2021-04-12 10:40:34 +00:00
presubmit_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test_mocks.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
pylintrc Undo enforcing of PEP-8 pylint changes for method and function names. 2020-11-10 18:26:25 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md Fix incorrect link in README.md 2021-04-20 10:58:08 +00:00
style-guide.md Enhance the readability of the style guide. 2021-04-23 15:35:56 +00:00
WATCHLISTS Add hta@ to rtc_base/ and api/ WATCHLISTS. 2021-01-06 09:43:34 +00:00
webrtc.gni sctp: Add DcsctpTransport based on dcSCTP 2021-05-06 09:38:49 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
whitespace.txt Reland "Triggering CI." 2021-03-22 11:57:23 +00:00

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