agilityfeat / webrtc-sip-exampleLinks
A small example of how to build a WebRTC application using SIP as signaling layer
☆35Updated 6 years ago
Alternatives and similar repositories for webrtc-sip-example
Users that are interested in webrtc-sip-example are comparing it to the libraries listed below
Sorting:
- React SIP user agent☆56Updated 6 years ago
- Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)☆44Updated 2 years ago
- Kamailio scripts to call from websocket UA to classic UA, and vice versa.☆44Updated 11 years ago
- UDP implementation of the SIP.js library☆38Updated 2 months ago
- ☆35Updated 4 years ago
- ☆62Updated 3 years ago
- Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network☆358Updated 5 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆70Updated this week
- Multiparty-meeting (mediasoup) SIP gateway using Kurento☆28Updated 6 years ago
- SIPREC recording server based on drachtio and rtpengine☆97Updated last year
- WebRTC SIP based VoIP client software (+chrome extension)☆111Updated last year
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆195Updated last week
- rtp record and rtp streamer☆72Updated 4 years ago
- tryit-jssip application☆88Updated 5 months ago
- ☆27Updated 8 years ago
- Example applications using SIP.js☆59Updated 8 years ago
- Node.js client and server for FreeSwitch Event Socket☆141Updated last year
- [POC] GStreamer plugin for Janus Gateway☆29Updated 2 years ago
- Javascript library to build a web-broswer softphone☆103Updated 6 months ago
- the open source SIP TelePresence system☆147Updated 6 years ago
- Media relay for RTP/RTCP and UDP streams☆47Updated 3 months ago
- Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf☆56Updated 5 months ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆308Updated 2 years ago
- Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration☆82Updated 10 years ago
- SIP/XMPP/WebRTC Application Server☆209Updated 3 weeks ago
- Load-balancing SIP proxy for Freeswitch☆28Updated 9 years ago
- Framework for functional and Load Testing of WebRTC☆55Updated 7 years ago
- Run janus gateway well configure for hublin in a Docker container.☆65Updated 5 years ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆37Updated 7 years ago
- nodejs client for accessing rtpengine via ng protocol☆25Updated 2 years ago