StefanYohansson / ktelephoneLinks
Qt softphone - Highly inspired by Telephone
☆21Updated 2 years ago
Alternatives and similar repositories for ktelephone
Users that are interested in ktelephone are comparing it to the libraries listed below
Sorting:
- Battle proven FreeSWITCH Event Socket Protocol client implementation with Gevent☆137Updated 2 months ago
- FreeSWITCH Event Socket Protocol client implementation with Elixir☆11Updated 3 months ago
- VoIP signaling and media test automation☆119Updated last month
- SIPssert Testing Framework is a tool used for facilitating conformity testing of complex VoIP setups☆61Updated 2 months ago
- Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification.☆298Updated 2 weeks ago
- SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console☆168Updated last month
- OpenSIPS CLI tool - an interactive command line tool that can be used to control and monitor OpenSIPS servers.☆95Updated last month
- Using Rust to write an asterisk module☆16Updated 11 years ago
- Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.☆187Updated 2 weeks ago
- FreeSWITCH Rust Bindings☆26Updated 8 years ago
- Various SIP resources.☆228Updated 10 months ago
- asyncio powered FreeSWITCH cluster control☆201Updated last year
- The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.☆443Updated last week
- SpanDSP is a low-level signal processing library that modulates and demodulates signals commonly used in telephony, such as the "noise" g…☆174Updated 2 weeks ago
- SIP performance test tool☆39Updated 2 years ago
- Foundational support for signalwire C products☆44Updated last week
- Modern and flexible SIP/VoIP cli tool☆350Updated 2 weeks ago
- Meta-repository to test interop between latest versions of opensips, kamailio, sippy b2bua and rtpproxy☆43Updated last month
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆172Updated last year
- Easier web calling by providing a layer of abstraction around SIP.js☆65Updated last year
- It's an open source restaurant and coffee shop management system.☆9Updated 2 years ago
- Sip Express Media Server☆170Updated 2 months ago
- Qt 5 port of Twinkle☆179Updated 7 months ago
- VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine☆124Updated 3 years ago
- Media relay for RTP/RTCP and UDP streams☆45Updated 2 months ago
- SIPp examples☆137Updated last year
- SIP framework built in Rust☆91Updated 3 years ago
- official docker images of kamailio project☆65Updated this week
- Web Admin Interface for Kamailio☆103Updated 4 months ago
- Baresip WebRTC Demo - moved to baresip☆47Updated 2 years ago