muggot / myphone3Links
☆25Updated 10 years ago
Alternatives and similar repositories for myphone3
Users that are interested in myphone3 are comparing it to the libraries listed below
Sorting:
- H323Plus - H.323 development framework☆112Updated 3 weeks ago
- BFCP message/client/server libraries☆13Updated 5 years ago
- Toolkit library for asynchronous network IO with protocol stacks including SIP, SDP, RTP, STUN, TURN, ICE, BFCP and DNS.☆41Updated 12 years ago
- Linphone.org mirror for mediastreamer2 (git://git.linphone.org/mediastreamer2.git)☆137Updated last week
- media sdk based on webrtc☆40Updated 2 years ago
- Examples of SIP register UA with sofia-sip, pjsip, libeXosip and libre☆29Updated 7 years ago
- Linphone.org mirror for belle-sip (git://git.linphone.org/belle-sip.git)☆73Updated 2 weeks ago
- WebRTC GIPS C/C++ API demos☆58Updated 11 years ago
- adaptive jitter buffer implementatioin☆32Updated 4 years ago
- rewrite WebRTC's jitter buffer for PJSIP☆14Updated 12 years ago
- oSIP is a LGPL implementation of SIP. It's stable, portable, flexible and compliant! -may be more-! It is used mostly with eXosip2 stack …☆44Updated 3 years ago
- Video Conferencing Server, based on openmcu-ru☆13Updated 11 years ago
- Audio and video processing media library☆97Updated 3 years ago
- SIP UserAgent(UAS and UAC) Sample☆38Updated 8 years ago
- Unofficial mirror/fork of http://svn.pjsip.org/repos/pjproject/trunk/ — check the Wiki for more information.☆54Updated 10 years ago
- the open source SIP TelePresence system☆149Updated 5 years ago
- sip video and voice client demo, receive rtsp media stream and push to other sip client. It use pjsip ,live555 and ffmpeg☆79Updated 9 years ago
- PTLib 2.10.9 fork for GNU Gatekeeper and H323Plus☆22Updated 3 weeks ago
- A sip server and client using pjsua2☆15Updated 5 years ago
- webrtc network part for reliable udp transport☆31Updated 4 years ago
- a gb28181 sip sdk based on resiprocate☆30Updated 6 years ago
- Linphone.org mirror for sofia-sip (git://git.linphone.org/sofia-sip.git)☆30Updated 7 months ago
- Implementation of RFC5766 and RFC5389☆77Updated 7 years ago
- WebRTC native C/C++ sdk api based release M67, just keep WebRTC's audio/video en/decode and transfer.☆22Updated 5 years ago
- Linphone.org mirror for bctoolbox (git://git.linphone.org/bctoolbox.git)☆25Updated 2 weeks ago
- Linphone.org mirror for bcg729 (git://git.linphone.org/bcg729.git)☆128Updated last year
- CSipSimple 是个款通用的支持SIP协议的互联网电话软件,可以在支持andriod的平板,手机上使用。支持语音编码: G.711 aLaw/uLaw, G.722.1, G.722, SPEEX, SPEEX-WB, AMR-WB, GSM, iLBC, G.729…☆60Updated 5 years ago
- C++ SIP stack based on Chromium source code☆24Updated 6 years ago
- 非常轻型的TS和PS封装与解封装代码,严格遵循ISO/IEC 13818-1标准,扩展性好。☆46Updated 8 years ago
- Baresip WebRTC Demo - moved to baresip☆47Updated 2 years ago