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.
☆173Updated 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☆211Updated last week
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆126Updated 3 years ago
- IVOZ Provider - Multitenant solution for VoIP telephony providers☆207Updated this week
- SIPREC recording server based on drachtio and rtpengine☆93Updated last year
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆111Updated last year
- Barebone Opensource Powered SBC☆110Updated 3 weeks ago
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆297Updated last year
- Various SIP resources.☆235Updated 3 weeks ago
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects☆165Updated last month
- An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.☆418Updated 2 months ago
- VoIP signaling and media test automation☆119Updated last month
- Web Admin Interface for Kamailio☆103Updated 5 months ago
- Sip Express Media Server☆172Updated 3 months ago
- Voip Open Linear Testing Suite☆43Updated last week
- Minimalist Windows / OSx / Linux SIP Softphone with API Control☆146Updated 4 months ago
- WebRTC SIP based VoIP client software (+chrome extension)☆108Updated 8 months ago
- A SIP call processing server that can be controlled via nodejs applications☆286Updated 3 weeks ago
- Scripts and such for Kamailio, Asterisk, FreeSWITCH, and more☆39Updated 2 weeks ago
- Modern and flexible SIP/VoIP cli tool☆353Updated 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
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/☆114Updated 2 years ago
- P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine☆104Updated 3 months ago
- Asterisk Management Interface (AMI) to Web-socket proxy☆87Updated 2 years ago
- ☆63Updated 3 years ago
- Stream Asterisk audio over Websockets☆182Updated 11 months ago
- Simple config file of Kamailio as Loadblancer for calls and registrations☆32Updated 3 years ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆174Updated 2 weeks ago
- API server and Web GUI for FreeSwitch written in Golang and Angular☆80Updated 4 months ago
- ☆35Updated last week
- HEP Capture Server for HOMER☆197Updated 2 weeks ago