gmaruzz / saraphoneLinks
SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. SaraPhone is fully integrated with FusionPBX. Based on SIP.js, SaraPhone works with all WebRTC compliant servers: FreeSWITCH, Asterisk, OpenSIPS, Kamailio, etc. SaraPhone gets its name from Giovanni's wife, Sara.
☆183Updated last year
Alternatives and similar repositories for saraphone
Users that are interested in saraphone are comparing it to the libraries listed below
Sorting:
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆220Updated 2 weeks ago
- IVOZ Provider - Multitenant solution for VoIP telephony providers☆216Updated this week
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆130Updated 3 years ago
- SIPREC recording server based on drachtio and rtpengine☆95Updated last year
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆112Updated last year
- Barebone Opensource Powered SBC☆113Updated 5 months ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆308Updated 2 years ago
- Various SIP resources.☆250Updated 3 months ago
- Web Admin Interface for Kamailio☆102Updated 10 months ago
- WebRTC SIP based VoIP client software (+chrome extension)☆111Updated last year
- Sip Express Media Server☆180Updated last month
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects☆168Updated 6 months ago
- An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.☆447Updated this week
- VoIP signaling and media test automation☆123Updated 2 months ago
- A SIP call processing server that can be controlled via nodejs applications☆297Updated this week
- Voip Open Linear Testing Suite☆44Updated 3 months ago
- Modern and flexible SIP/VoIP cli tool☆368Updated 3 weeks ago
- Minimalist Windows / OSx / Linux SIP Softphone with API Control☆152Updated 9 months ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆179Updated 5 months ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆37Updated 7 years ago
- 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 …☆332Updated last year
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆40Updated 2 months ago
- Simple config file of Kamailio as Loadblancer for calls and registrations☆31Updated 4 years ago
- Asterisk Management Interface (AMI) to Web-socket proxy☆90Updated 2 years ago
- SIP Phone WebRTC for your browser☆59Updated 4 years ago
- Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCH☆56Updated 2 years ago
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/☆123Updated 2 years ago
- ☆36Updated last week
- Stream Asterisk audio over Websockets☆186Updated 3 months ago
- API server and Web GUI for FreeSwitch written in Golang and Angular☆97Updated last week