collecttix / ctxSipLinks
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 as a popup from within your application. Works well with Kazoo from 2600hz
☆325Updated last year
Alternatives and similar repositories for ctxSip
Users that are interested in ctxSip are comparing it to the libraries listed below
Sorting:
- Asterisk Management Interface (AMI) to Web-socket proxy☆87Updated 2 years ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆175Updated last year
- Setup for a WEBRTC client and Kamailio server to call SIP clients☆298Updated last year
- WebRTC SIP based VoIP client software (+chrome extension)☆108Updated 9 months ago
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects☆165Updated 2 months ago
- A SIP call processing server that can be controlled via nodejs applications☆286Updated last month
- Minimalist Windows / OSx / Linux SIP Softphone with API Control☆146Updated 5 months ago
- Web Admin Interface for Kamailio☆103Updated 6 months ago
- IVOZ Provider - Multitenant solution for VoIP telephony providers☆207Updated this week
- A fully featured browser based WebRTC SIP phone for Asterisk☆596Updated 9 months ago
- Stream Asterisk audio over Websockets☆182Updated 11 months ago
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆112Updated last year
- Javascript library to build a web-broswer softphone☆99Updated last month
- P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine☆104Updated 4 months ago
- ☆63Updated 3 years ago
- Dashboard for Queues in Asterisk and FreeSWITCH. app_queue panel for Asterisk and mod_callcenter in FreeSWITCH. Get news -> http://eepu…☆185Updated last year
- SIP Phone WebRTC for your browser☆60Updated 4 years ago
- SIPREC recording server based on drachtio and rtpengine☆94Updated last year
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆211Updated last week
- Barebone Opensource Powered SBC☆110Updated last month
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆128Updated 3 years ago
- FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications☆176Updated 2 weeks ago
- React based front-end demo for Asterisk's SFU☆30Updated 2 years ago
- Easier web calling by providing a layer of abstraction around SIP.js☆65Updated last year
- New tryit-jssip application☆88Updated last year
- The world's first HTML5 SIP client (WebRTC)☆955Updated 4 years ago
- Node.js client and server for FreeSwitch Event Socket☆138Updated 8 months ago
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆190Updated 2 months ago
- Asterisk AGI script that makes use of Microsoft Azure Cognitive Services for text to speech synthesis☆19Updated 3 years ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆175Updated last month