OpenTelecom / react-sip-phone
OTF React SIP.JS Phone
☆23Updated last year
Alternatives and similar repositories for react-sip-phone:
Users that are interested in react-sip-phone are comparing it to the libraries listed below
- Core telephony feature server for the jambones platform☆59Updated this week
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆169Updated 10 months ago
- Classes of Telephony for the Python's diagrams package☆11Updated 3 years ago
- React SIP user agent☆53Updated 5 years ago
- A WebRTC Phone For Vicidial☆32Updated 2 years ago
- Multitenant PBX☆30Updated last year
- Simple config file of Kamailio as Loadblancer for calls and registrations☆32Updated 3 years ago
- Wazo Platform SBC Kamailio configuration for the C4 infrastructure☆13Updated 2 years ago
- SIPp Call Scenario for Performance Test☆17Updated 4 years ago
- SIPREC recording server based on drachtio and rtpengine☆89Updated 9 months ago
- P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine☆103Updated last month
- SMS, Fax, Voice Broadcasting and auto dialer Software, A unified communications open source autodialer developed over freeswitch communic…☆88Updated last month
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆107Updated last year
- ☆35Updated last month
- ☆77Updated last month
- ☆62Updated 2 years ago
- Enterprise telephony recording and retrieval system with web based user interface.☆29Updated 3 years ago
- The Asterisk Documentation Project.☆37Updated this week
- SIP Phone WebRTC for your browser☆59Updated 4 years ago
- Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://jira.astppbilling.org☆160Updated last week
- Mobile push notifications for RTC infrastructures☆15Updated 11 months ago
- Voip Open Linear Testing Suite☆42Updated 2 weeks ago
- ☆14Updated 2 years ago
- Asterisk based call center solution☆54Updated 6 years ago
- WebRTC SIP based VoIP client software (+chrome extension)☆106Updated 5 months ago
- WebRTC based webphone for Vicidial☆35Updated 2 years ago
- Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library☆50Updated 4 months ago
- Asterisk AGI script that makes use of Microsoft Azure Cognitive Services for text to speech synthesis☆19Updated 2 years ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆119Updated 3 years ago
- Freeswitch setup, profiles , dial-plans and vars for various use-cases☆17Updated 3 years ago