InnovateAsterisk / Browser-PhoneLinks
A fully featured browser based WebRTC SIP phone for Asterisk
☆616Updated 11 months ago
Alternatives and similar repositories for Browser-Phone
Users that are interested in Browser-Phone 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…☆178Updated last year
- An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.☆436Updated last month
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆307Updated 2 years ago
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆218Updated 2 weeks 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
- IVOZ Provider - Multitenant solution for VoIP telephony providers☆210Updated last week
- A SIP call processing server that can be controlled via nodejs applications☆291Updated 3 months ago
- Various SIP resources.☆243Updated last month
- Stream Asterisk audio over Websockets☆186Updated 2 months ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆130Updated 3 years ago
- Barebone Opensource Powered SBC☆110Updated 3 months ago
- Official FusionPBX - A full-featured domain based multi-tenant PBX and voice switch for FreeSwitch.☆920Updated this week
- Modern and flexible SIP/VoIP cli tool☆363Updated this week
- SIP Phone WebRTC for your browser☆59Updated 4 years ago
- ☆87Updated 7 months ago
- The Sipwise media proxy for Kamailio☆882Updated last week
- Docker image providing Asterisk PBX☆262Updated 4 months ago
- Dashboard for Queues in Asterisk and FreeSWITCH. app_queue panel for Asterisk and mod_callcenter in FreeSWITCH. Get news -> http://eepu…☆187Updated last year
- drachtio signaling resource framework☆198Updated last week
- Node.js client for ARI. This library is best effort with limited support.☆274Updated last year
- Kamailio Tutorial Examples from Blog @ https://nickvsnetworking.com/category/voip/kamailio/☆123Updated 2 years ago
- Voip Open Linear Testing Suite☆43Updated last month
- SIPp scenarios I use for testing SIP stuff☆300Updated last year
- SIPREC recording server based on drachtio and rtpengine☆95Updated last year
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆111Updated last year
- Demo of scalable Asterisk on Kubernetes☆173Updated 2 years ago
- WebRTC SIP based VoIP client software (+chrome extension)☆110Updated 11 months ago
- Minimalist Windows / OSx / Linux SIP Softphone with API Control☆150Updated 7 months ago
- ☆19Updated last year
- VoIP signaling and media test automation☆121Updated last week