vanbui1995 / react-sipjsLinks
React components for SIP.js
☆25Updated last year
Alternatives and similar repositories for react-sipjs
Users that are interested in react-sipjs are comparing it to the libraries listed below
Sorting:
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆183Updated last year
- ☆90Updated this week
- A SIP call processing server that can be controlled via nodejs applications☆305Updated 3 weeks ago
- Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCH☆56Updated 3 years ago
- SIP Phone WebRTC for your browser☆59Updated 4 years ago
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆225Updated last week
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆308Updated 2 years ago
- Core telephony feature server for the jambones platform☆88Updated this week
- Stream Asterisk audio over Websockets☆188Updated 5 months ago
- OpenSIPS AI Voice Connector Community Edition Platform☆71Updated 2 months ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆130Updated last week
- ctxSip is a Javascript based SIP client that uses WebRTC and WebSockets to connect to your SIP server. The UI is designed to be launched …☆336Updated last year
- Teams Direct Routing SBC connects your local PBX with MS Teams voice platform.☆12Updated last year
- Wazo Platform SBC Kamailio configuration for the C4 infrastructure☆13Updated 3 years ago
- Asterisk with PostgreSQL Real-time Database☆23Updated 5 years ago
- ☆16Updated 3 years ago
- Kamailio in docker container with TLS enabled using Let's Encrypt.☆25Updated 3 years ago
- Simple config file of Kamailio as Loadblancer for calls and registrations☆31Updated 4 years ago
- P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine☆109Updated last month
- Barebone Opensource Powered SBC☆114Updated 6 months ago
- WebRTC SIP based VoIP client software (+chrome extension)☆111Updated last year
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆114Updated 3 weeks ago
- Call API is a front-end layer for managing advanced SIP call flows. It listens for WebSocket connections and talks JSON-RPC 2.0 over them…☆56Updated last month
- Kubernetes dynamic configuration engine for Asterisk☆71Updated 3 years ago
- Asterisk Management Interface (AMI) to Web-socket proxy☆92Updated 2 years ago
- Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf☆56Updated 4 months ago
- SIPREC recording server based on drachtio and rtpengine☆96Updated last year
- Web Admin Interface for Kamailio☆103Updated 11 months ago
- IVOZ Provider - Multitenant solution for VoIP telephony providers☆218Updated last week
- Call Analytics Solution for Freeswitch, Asterisk, Kamailio and other VoIP Switches☆305Updated 3 years ago