No description
Find a file
Philipp Hancke 21c4b1e9ca stats: expose relayProtocol on prflx candidate
This makes relay candidates identifiable in scenarios
where the TURN server is behind another entity and the peer
sees a different ip address:

client -> turn -> relay address -> third party relay + address -> peer

In those cases, the relay candidate will become peer-reflexive
since the peer sends the third party relay's address in the xor-mapped
address and it is currently not easily possible to determine this is a
relay candidate anymore.

BUG=webrtc:13392

Change-Id: I6787339d0abdc735f8a43f636a676cccd8cadcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237561
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35346}
2021-11-15 17:14:29 +00:00
api Implement RTCOutboundRtpStreamStats.targetBitrate for audio. 2021-11-12 09:24:34 +00:00
audio Implement RTCOutboundRtpStreamStats.targetBitrate for audio. 2021-11-12 09:24:34 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Remove usage of INFO alias for LS_INFO in log messages 2021-11-04 13:46:17 +00:00
common_audio Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
common_video Remove unused IXXXBuffer::PasteFrom 2021-10-26 11:55:31 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Add contributing.md 2021-11-03 14:59:46 +00:00
examples Remove usage of INFO alias for LS_INFO in log messages 2021-11-04 13:46:17 +00:00
g3doc Add static AsString functions for PeerConnectionInterface enums 2021-11-02 12:29:50 +00:00
logging Remove usage of INFO alias for LS_INFO in log messages 2021-11-04 13:46:17 +00:00
media Implement RTCOutboundRtpStreamStats.targetBitrate for audio. 2021-11-12 09:24:34 +00:00
modules Correctly set first/last packet of frame bit in VideoRtpDepacketizerVp9. 2021-11-15 16:22:09 +00:00
net/dcsctp dcsctp: Use strong type for MaxRetransmits 2021-11-08 20:14:15 +00:00
p2p Revert "Remove code supporting the SDES crypto mode in SDP" 2021-11-04 14:46:27 +00:00
pc stats: expose relayProtocol on prflx candidate 2021-11-15 17:14:29 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Permit current queue reference to be null on sequence checker creation 2021-11-15 09:39:10 +00:00
rtc_tools Remove usage of INFO alias for LS_INFO in log messages 2021-11-04 13:46:17 +00:00
sdk Request DTMF sender only for audio sender in iOS SDK. 2021-11-08 18:07:35 +00:00
stats Implement missing candidate pair packets/bytes sent/received stats. 2021-09-28 23:27:05 +00:00
system_wrappers Removed timezone usage in UnixRealTimeClock::CurrentTimeVal when calling gettimeofday. 2021-11-02 05:32:10 +00:00
test Remove usage of INFO alias for LS_INFO in log messages 2021-11-04 13:46:17 +00:00
tools_webrtc Disable DCHECKs on sanitizer builds. 2021-10-31 13:17:38 +00:00
video Correctly set first/last packet of frame bit in VideoRtpDepacketizerVp9. 2021-11-15 16:22:09 +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 Increase iOS deployment target from 10 to 12. 2021-07-02 17:02:27 +00:00
.vpython Update links to point at main branch 2021-07-22 16:41:26 +00:00
.vpython3 Add .vpython3 file to webrtc 2021-09-15 16:56:30 +00:00
AUTHORS Added main profile to supported H264 codecs 2021-11-10 10:56:13 +00:00
BUILD.gn Add GN arg to force RTC_DLOG to be ON. 2021-10-12 08:48:22 +00:00
CODE_OF_CONDUCT.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +00:00
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 Remove infra/tools/luci/isolated from DEPS. 2021-11-15 08:44:43 +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 phoglund from ENG_REVIEW_OWNERS 2021-10-08 08:29:42 +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 Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +00:00
OWNERS Add .vpython3 file to webrtc 2021-09-15 16:56:30 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py fix some typos 2021-08-12 18:37:10 +00:00
presubmit_test.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +00:00
presubmit_test_mocks.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +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 doc: add g3doc sitemap to toplevel readme 2021-07-23 07:55:17 +00:00
WATCHLISTS Update WATCHLISTS 2021-08-23 13:37:55 +00:00
webrtc.gni Fix typo in comment. 2021-10-12 15:10:50 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Test LKGR finder. 2021-09-24 20:09:34 +00:00

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