baresip is a portable SIP user agent designed as a minimal and modular VoIP client.
It supports Linux and other platforms and is suitable for lightweight softphone use, embedded deployments, and custom real-time communication applications.
This is free and open source software.
Key Features
- Modular plugin architecture with modules for codecs, transports, audio and video backends, interfaces, and protocol extensions.
- Comprehensive SIP telephony features including multiple accounts and calls, call transfer, hold and resume, conferencing, messaging, redial, and presence-aware address book support.
- Broad media support with audio codecs such as Opus, G.711, G.722, G.726, AAC, and Codec2, plus video codecs including H.264, H.265, VP8, VP9, and AV1.
- Security and NAT traversal features including TLS, Secure RTP, DTLS-SRTP, ZRTP, STUN, TURN, ICE, NAT-PMP, and PCP.
- Multiple Linux-friendly audio and video backends including ALSA, PulseAudio, JACK, GStreamer, Video4Linux2, SDL2, and X11.
- Simple configuration files together with command-line and network console interfaces, plus an embedded HTTP server for management and integration.
Website: github.com/baresip/baresip
Support:
Developer: Alfred E. Heggestad and contributors
License: BSD 3-Clause License
Baresip is written in C. Learn C with our recommended free books and free tutorials.
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Read our verdict in the software roundup.
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