saltyrtc / saltyrtc-demoLinks
Small SaltyRTC demo application with a web interface and an Android app.
☆28Updated 5 years ago
Alternatives and similar repositories for saltyrtc-demo
Users that are interested in saltyrtc-demo are comparing it to the libraries listed below
Sorting:
- SaltyRTC signalling server implementation.☆60Updated 3 years ago
- Sylk WebRTC client☆60Updated last month
- Audio & Video chat for Etherpad - Video Conferencing with a focus on collaboration☆75Updated last month
- Jitsi Autoscaler microservice☆30Updated 2 weeks ago
- Allows MediaStream to switch tracks without setting srcObject this allows MediaRecording to continue recording☆31Updated 4 years ago
- WebRTC CodeLab☆15Updated 8 months ago
- Pipe WebRTC MediaStreams to/from FFMPEG.☆88Updated 4 years ago
- Javascript library to build a web-broswer softphone☆99Updated this week
- A traffic generator for Jitsi Videobridge☆61Updated 5 years ago
- ☆37Updated 3 years ago
- SFrame.js pure javascript implementation based on webcrypto☆51Updated 2 years ago
- Control server for WebRTC SFU☆52Updated this week
- JavaScript SylkRTC API library☆29Updated last month
- Sofia sip stack (forked from gitorious 1/1/2014)☆18Updated 4 months ago
- MediaSources from Readable streams☆23Updated 6 years ago
- Online Conference and Collaboration Tool☆65Updated last year
- Physical meetings rooms, reimagined.☆73Updated last week
- P2P Screen Sharing with WebRTC☆88Updated 5 years ago
- Opentok app with screen sharing using the WebRTC screen sharing feature☆40Updated 2 years ago
- A small bash script for installing and configuring coturn, kurento and asterisk on Ubuntu 16.04☆12Updated 7 years ago
- Conference Lightweight Bridging Javascript Implementation☆41Updated 4 years ago
- The world's first Open Source Emergency Warning System Dissemination Platform☆30Updated this week
- [ARCHIVED] Contents migrated to monorepo: https://github.com/Kurento/kurento☆48Updated 2 years ago
- Open Source Telephony API Platform☆88Updated last year
- Multi-User Video Conference☆59Updated 12 years ago
- SIP gateway for Jigasi based on FreeSWITCH☆10Updated last month
- Basic meeting webservice (client + server) based on WebRTC technology☆20Updated 7 years ago
- WebRTC interoperability tests☆27Updated 2 years ago
- Web Audio and P2P Call for client side audio conferencing☆14Updated 9 years ago
- Protocol description and organisational information for SaltyRTC implementations.☆77Updated 5 years ago