yjhui0331 Goto Github PK
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yjhui0331's Projects
mediasoup
Cutting Edge WebRTC Video Conferencing
mixtonephone
mod_gb28181
Freeswitch gb28181 module
mymediaplayer
netty-websocket
a fully-functioning websocket server built on netty.
nodejs_call_c_cc
nodejs call c/c++
openonvif
Free ONVIF library.
openrtmp
owt-server
General server (streaming/conference/transcoding/anayltics) for OWT. (A.k.a. MediaServer)
peertc
peer to peer message and file sender
pjsip
pjsip study
pjsipclient
SIP Client with PJSIP - Use with ASTERISK Server
player-cpp-ffmpeg-sdl
SDL2 and FFmpeg Video Player in C++
qt-onvif-client
qt实现的onvif的客户端
rapidonvif
ONVIF C++ lib, support ProfileS, mail to [email protected] for latest version
rtc_gw
Webrtc Gateway to SIP/RTP (experimental work in progress)
rtp
Implementation of a RTP server that sends video stream (H.264/HEVC) using the Real-time Transport Protocol(RTP) based on Linux/MacOS. 一个基于Linux/MacOS平台的可以发送携带H.264/HEVC媒体类型的RTP视频流的示例程序。
rtpengine
The Sipwise media proxy for Kamailio
rtspclientlib
RtspClient with live555
rtsplibrary
sdk
portable system call(aio/socket/thread/process/lock/event/pipe)
sip2rtsp
sip/ims protocol to rtsp protocol gateway
sip_stacks_examples
Examples of SIP register UA with sofia-sip, pjsip, libeXosip and libre
sip_ua
SIP UserAgent(UAS and UAC) Sample
sipc
A lightweight SIP client written in ANSI C.
sipclient
sip client，gb28181 client
sipml5-demo
a SIP client demo based on sipML5
sipproject
siprtcproxy
网关服务：Sip与Rtc互通，实现Web，Android，iOS，小程序，SIP座机，PSTN电话，手机互通。
sipsdk
a gb28181 sip sdk based on resiprocate
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