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.
☆175Updated 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☆215Updated last week
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆128Updated 3 years ago
- Barebone Opensource Powered SBC☆110Updated last month
- IVOZ Provider - Multitenant solution for VoIP telephony providers☆207Updated last week
- SIPREC recording server based on drachtio and rtpengine☆94Updated last year
- Various SIP resources.☆237Updated last month
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆112Updated last year
- VoIP signaling and media test automation☆119Updated last month
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆298Updated last year
- Minimalist Windows / OSx / Linux SIP Softphone with API Control☆147Updated 5 months ago
- A SIP call processing server that can be controlled via nodejs applications☆286Updated last month
- WebRTC SIP based VoIP client software (+chrome extension)☆108Updated 9 months ago
- An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.☆423Updated 2 weeks ago
- Voip Open Linear Testing Suite☆43Updated 3 weeks ago
- Modern and flexible SIP/VoIP cli tool☆356Updated last week
- Sip Express Media Server☆172Updated 3 months ago
- Web Admin Interface for Kamailio☆103Updated 6 months ago
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects☆165Updated 2 months ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆175Updated last month
- 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 …☆325Updated last year
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 7 years ago
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/☆118Updated 2 years ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆39Updated last month
- Stream Asterisk audio over Websockets☆182Updated last year
- Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCH☆55Updated 2 years ago
- Asterisk Management Interface (AMI) to Web-socket proxy☆87Updated 2 years ago
- HEP Capture Server for HOMER☆198Updated last month
- SIP performance test tool☆39Updated 2 years ago
- ☆35Updated 3 weeks ago
- Kamailio Command Line Interface Control Tool☆56Updated 7 months ago