anthmFS / FreeSWITCH
FreeSWITCH
☆27Updated 4 years ago
Related projects: ⓘ
- Asterisk module that provides the "eSpeak" dialplan application. It allows you to use the eSpeak text to speech synthesizer. Works with a…☆41Updated 7 months ago
- ORTC Community Group specification repository (see W3C WebRTC for official standards track)☆122Updated last year
- ☆88Updated 4 years ago
- OverSIP: the SIP framework you dreamed about☆340Updated 4 years ago
- WebRTC library to make media management easier across different browsers.☆73Updated 7 years ago
- TESTING - replicated hourly from Google Code SVN using sync2git☆58Updated 8 years ago
- Fork of the Asterisk VOIP software with support for the Opus codec☆14Updated 12 years ago
- FreeSWITCH mirror (not automatically updated)☆27Updated 10 years ago
- the open source SIP TelePresence system☆148Updated 4 years ago
- A generic Jingle session manager implementation, suitable for integration by other XMPP libraries.☆89Updated 4 years ago
- JavaScript SylkRTC API library☆29Updated 8 months ago
- YETI application for SEMS core☆16Updated 2 weeks ago
- Conference Lightweight Bridging Javascript Implementation☆42Updated 3 years ago
- Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network☆342Updated 4 years ago
- A parser/serializer for SDP to JSON. Useful for converting SDP to other formats like Jingle for WebRTC signalling☆41Updated last year
- A Python Wrapper to WebRTC project☆34Updated 9 years ago
- Scripts from the 2600hz Community to help manage, enhance and debug the KAZOO platform☆41Updated 11 months ago
- Voice quality probe for end-to-end measurement of jitter and packet loss☆40Updated 9 years ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆76Updated 9 years ago
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆170Updated 3 weeks ago
- SIP in JavaScript☆46Updated 8 years ago
- Web status monitor for FreeSWITCH's mod_callcenter queues and agents☆38Updated 10 years ago
- webrtc connection plugin for strophe.js☆156Updated 9 years ago
- Verto jQuery library documentation.☆53Updated 4 years ago
- Sylk WebRTC client☆59Updated this week
- First Open Source Billing Platform for FreeSWITCH☆49Updated 10 years ago
- RTPProxy - application for RTP packets relaying - additional patches☆28Updated 10 years ago
- Multi-User Video Conference☆59Updated 11 years ago
- Asterisk module that provides the "Flite" dialplan application, which allows you to use the Flite text to speech engine with Asterisk..Wo…☆28Updated last year
- My online writings. This time its about OpenSIPS 101☆28Updated 9 years ago