AGProjects / openxcap
Fully featured XCAP server (RFC 4825)
☆19Updated last week
Alternatives and similar repositories for openxcap
Users that are interested in openxcap are comparing it to the libraries listed below
Sorting:
- Fork of the Asterisk VOIP software with support for the Opus codec☆14Updated 12 years ago
- Meta-repository to test interop between latest versions of opensips, kamailio, sippy b2bua and rtpproxy☆43Updated this week
- SIP Express Media Server, very fast and flexible SIP media server☆65Updated last week
- Examples of SIP register UA with sofia-sip, pjsip, libeXosip and libre☆28Updated 7 years ago
- zmq/json support asterisk AMI module. (zeromq, 0MQ)☆16Updated last year
- Secure Telephone Identity Management in Session Initiation Protocol☆8Updated 8 years ago
- Asterisk module that provides the "eSpeak" dialplan application. It allows you to use the eSpeak text to speech synthesizer. Works with a…☆42Updated last year
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆79Updated 10 years ago
- FreeSWITCH admin panel☆23Updated 13 years ago
- SIP Voice Quality reports Collector☆14Updated 8 years ago
- Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS☆13Updated 10 years ago
- C library implementing STIR-shaken STI-SP AS/VS, STI-CA☆31Updated 8 months ago
- Restcomm Session Border Controller☆24Updated 6 years ago
- YETI application for SEMS core☆16Updated 3 weeks ago
- JavaScript SylkRTC API library☆29Updated 2 weeks ago
- Linphone.org mirror for bctoolbox (git://git.linphone.org/bctoolbox.git)☆25Updated last week
- standalone audio conferencing server based on resiprocate/recon 1.8 release series☆14Updated 7 years ago
- Linphone.org mirror for belle-sip (git://git.linphone.org/belle-sip.git)☆72Updated 5 months ago
- Media relay for RTP/RTCP and UDP streams☆45Updated last month
- Yet Another Telephony Engine - UNOFFICIAL mirror☆114Updated 6 years ago
- Intel IPP audio codecs including G.729 and G.723.1 adapted for Asterisk☆60Updated 12 years ago
- Documentation and Tutorials for Kamailio SIP Server☆28Updated 2 weeks ago
- Web application to faciliate benchmarking and testing SIP based services☆19Updated 9 years ago
- Audio fingerprinting and recognition module for the Asterisk☆10Updated 6 years ago
- Fixed version of Kamailio IMS configuration files for basic calling☆51Updated 11 months ago
- Sofia sip stack (forked from gitorious 1/1/2014)☆18Updated 4 months ago
- sprout and bono, the Clearwater SIP router and edge proxy☆40Updated 5 years ago
- Real-time multi-modal inference server for interacting with humans and other intelligences around☆17Updated 4 months ago
- complete SIP signalling and RTP media service for rapid development of voice/video services and softphones☆34Updated 4 months ago
- Enterprise telephony recording and retrieval system with web based user interface.☆29Updated 3 years ago