Open Source VoIP Softphones

Opus is a lossy audio compression standard and codec designed for high-quality speech and music transmission over the internet. It functions as a low-latency audio codec and network-resilient streamer, providing a framework for encoding and decoding digital audio. The project distinguishes itself through the support of multi-channel ambisonics for immersive three-dimensional spatial audio reproduction. It is specifically optimized for real-time interactive communication, utilizing dynamic bitrate adjustment and forward error correction to maintain audio quality on unstable networks. The syst

FreeSWITCH is an open-source software telephony switch that routes and processes voice, video, and messaging calls over IP networks using standard protocols such as SIP and RTP. It serves as both a call routing and switching engine and a programmable telephony API, enabling developers to build custom telephony applications with event-driven call control and media manipulation. As a SIP media server, it handles media streaming, transcoding, and conferencing, and functions as a unified communications platform integrating voice, video, messaging, and conferencing into a single switchable infrastr

Coturn is a network server that facilitates peer-to-peer media traffic for real-time communication applications. It functions as a relay platform for voice, video, and data transmission, enabling direct connections between clients located behind restrictive firewalls and network address translators. The server implements standard network traversal protocols to manage media packet exchange and client authentication. It utilizes a multi-threaded architecture and event-driven polling to handle high-throughput traffic, while employing hash-based message authentication codes to verify client ident