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Victor Boivie d68f18ee6e dcsctp: Allow specifying minimum RTT variance 
This is mentioned in
https://datatracker.ietf.org/doc/html/rfc4960#section-6.3.1 and further
described in https://datatracker.ietf.org/doc/html/rfc6298#section-4.

The TCP RFCs mentioned G as the clock granularity, but in SCTP it should
be set much higher, to account for the delayed ack timeout (ATO) of the
peer (as that can be seen as a very high clock granularity). That one is
set to 200ms by default in many clients, so a reasonable default limit
could be set to 220 ms.

If the measured variance is higher, it will be used instead.

Bug: webrtc:12943
Change-Id: Ifc217daa390850520da8b3beb0ef214181ff8c4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232614
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35068}
2021-09-22 20:22:54 +00:00
api Mark toI420 as Nullable 2021-09-21 10:05:09 +00:00
audio Deduplicate set of the rtp header extension uri constants 2021-09-14 13:38:44 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Update WebRTC code version (2021-09-22T04:04:23). 2021-09-22 05:58:49 +00:00
common_audio Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
common_video Fix integer overflow in h264 pps parser 2021-09-20 11:28:36 +00:00
data
docs Update Mac prerequisites 2021-08-23 19:52:17 +00:00
examples Replace AV1X with AV1 2021-09-14 08:29:02 +00:00
g3doc Update missed link to chromium C++11 styleguide 2021-09-20 21:05:39 +00:00
logging Migrate rtc event log from rtc::BitBuffer to BitstreamReader 2021-09-07 14:03:27 +00:00
media sctp: dcsctp: Manage lifecycle explicitly 2021-09-22 07:17:29 +00:00
modules rtp_rtcp: use webrtc::flat_map for remote_senders 2021-09-21 13:20:16 +00:00
net/dcsctp dcsctp: Allow specifying minimum RTT variance 2021-09-22 20:22:54 +00:00
p2p Delete BasicPacketSocketFactory constructor with thread argument 2021-09-22 12:15:06 +00:00
pc Delete BasicPacketSocketFactory constructor with thread argument 2021-09-22 12:15:06 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Delete BitBuffer 2021-09-21 16:28:38 +00:00
rtc_tools Delete BasicPacketSocketFactory constructor with thread argument 2021-09-22 12:15:06 +00:00
sdk [ios] Fix two rtc_unittests that fail when using lld as linker 2021-09-22 19:45:44 +00:00
stats Reland "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-06 14:26:55 +00:00
system_wrappers Use GTEST_SKIP() instead of early return. 2021-08-12 15:24:13 +00:00
test In vp9 encoder fuzzer exclude testing unsupported bitrate configurations 2021-09-22 16:20:56 +00:00
tools_webrtc rename use_x11 to ozone_platform_x11 2021-09-20 15:13:55 +00:00
video New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. 2021-09-15 09:57:29 +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 rename use_x11 to ozone_platform_x11 2021-09-20 15:13:55 +00:00
BUILD.gn Allow export of Obj-C symbols without C++ ones. 2021-07-30 22:54:59 +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 Roll chromium_revision 39bb5dc28c..b32d012bf3 (923768:923877) 2021-09-22 16:47:26 +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
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
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 rename use_x11 to ozone_platform_x11 2021-09-20 15:13:55 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger bots 2021-09-08 18:30:03 +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