open-voip-alliance / WebphoneLibLinks
Easier web calling by providing a layer of abstraction around SIP.js
☆65Updated last year
Alternatives and similar repositories for WebphoneLib
Users that are interested in WebphoneLib 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
- IVOZ Provider - Multitenant solution for VoIP telephony providers☆210Updated last week
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆218Updated 2 weeks ago
- VoIP signaling and media test automation☆121Updated last week
- P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine☆106Updated 6 months ago
- React based front-end demo for Asterisk's SFU☆30Updated 2 years ago
- SIPREC recording server based on drachtio and rtpengine☆95Updated last year
- 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
- Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf☆55Updated last month
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆130Updated 3 years ago
- A SIP call processing server that can be controlled via nodejs applications☆291Updated 3 months ago
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects☆166Updated 4 months ago
- Barebone Opensource Powered SBC☆110Updated 3 months ago
- ☆36Updated last week
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆307Updated 2 years ago
- ☆87Updated 7 months ago
- WebRTC SIP based VoIP client software (+chrome extension)☆110Updated 11 months ago
- Javascript library to build a web-broswer softphone☆101Updated 3 months ago
- Load-balancing SIP proxy for Freeswitch☆26Updated 9 years ago
- REST API for sharing IP addresses sending unwanted SIP traffic☆62Updated last year
- Ari-proxy connects Asterisk, an open source communication server, to the Apache Kafka distributed streaming platform.☆62Updated last week
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆37Updated 7 years ago
- Real-time Charging System for Telecom & ISP environments☆471Updated last week
- Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCH☆56Updated 2 years ago
- Modern and flexible SIP/VoIP cli tool☆363Updated this week
- 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
- Battle proven FreeSWITCH Event Socket Protocol client implementation with Gevent☆144Updated 6 months ago
- SIPssert Testing Framework is a tool used for facilitating conformity testing of complex VoIP setups☆63Updated 2 months ago
- ☆24Updated last year