GitHub - UWNetworksLab/webrtc-mod: A fork of WebRTC used to support UDP shapeshifting in uProxy for mobile devices

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A fork of WebRTC used to support UDP shapeshifting in uProxy for mobile devices

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Name		Name	Last commit message	Last commit date
Latest commit History 5,721 Commits
api		api
audio		audio
base		base
build		build
call		call
common_audio		common_audio
common_video		common_video
examples		examples
libjingle		libjingle
media		media
modules		modules
p2p		p2p
pc		pc
sound		sound
system_wrappers		system_wrappers
test		test
tools		tools
video		video
voice_engine		voice_engine
.gitignore		.gitignore
0001-bemasc-xwalk-hack-disable-STUN-parsing-check-in-UDP-.patch		0001-bemasc-xwalk-hack-disable-STUN-parsing-check-in-UDP-.patch
AUTHORS		AUTHORS
BUILD.gn		BUILD.gn
LICENSE		LICENSE
LICENSE_THIRD_PARTY		LICENSE_THIRD_PARTY
OWNERS		OWNERS
PATENTS		PATENTS
PRESUBMIT.py		PRESUBMIT.py
README.chromium		README.chromium
audio_receive_stream.h		audio_receive_stream.h
audio_send_stream.h		audio_send_stream.h
audio_state.h		audio_state.h
call.h		call.h
codereview.settings		codereview.settings
common.gyp		common.gyp
common.h		common.h
common_types.cc		common_types.cc
common_types.h		common_types.h
config.cc		config.cc
config.h		config.h
engine_configurations.h		engine_configurations.h
frame_callback.h		frame_callback.h
rtc_media_unittests.isolate		rtc_media_unittests.isolate
rtc_pc_unittests.isolate		rtc_pc_unittests.isolate
rtc_unittests.isolate		rtc_unittests.isolate
stream.h		stream.h
supplement.gypi		supplement.gypi
transport.h		transport.h
typedefs.h		typedefs.h
video_decoder.h		video_decoder.h
video_encoder.h		video_encoder.h
video_engine_tests.isolate		video_engine_tests.isolate
video_frame.h		video_frame.h
video_receive_stream.h		video_receive_stream.h
video_renderer.h		video_renderer.h
video_send_stream.h		video_send_stream.h
webrtc.gyp		webrtc.gyp
webrtc_examples.gyp		webrtc_examples.gyp
webrtc_nonparallel_tests.isolate		webrtc_nonparallel_tests.isolate
webrtc_perf_tests.isolate		webrtc_perf_tests.isolate
webrtc_tests.gypi		webrtc_tests.gypi

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Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.