# baresip **Repository Path**: iscar/baresip ## Basic Information - **Project Name**: baresip - **Description**: Baresip is a modular SIP User-Agent with audio and video support - **Primary Language**: Unknown - **License**: BSD-3-Clause - **Default Branch**: main - **Homepage**: None - **GVP Project**: No ## Statistics - **Stars**: 0 - **Forks**: 2 - **Created**: 2025-11-14 - **Last Updated**: 2025-11-14 ## Categories & Tags **Categories**: Uncategorized **Tags**: None ## README baresip README ============== ![Baresip Logo](https://raw.githubusercontent.com/baresip/baresip/master/share/logo.png) Baresip is a portable and modular SIP User-Agent with audio and video support. Copyright (c) 2010 - 2025 Alfred E. Heggestad and Contributors Distributed under BSD license ![Build](https://github.com/baresip/baresip/workflows/Build/badge.svg) ![Lint](https://github.com/baresip/baresip/workflows/lint/badge.svg) ![OpenSSL and LibreSSL](https://github.com/baresip/baresip/workflows/OpenSSL%20no-deprecated%20and%20LibreSSL/badge.svg) ![Valgrind](https://github.com/baresip/baresip/workflows/valgrind%20leak%20check/badge.svg) ## Features: * Call features: - Unlimited number of SIP accounts - Unlimited number of calls - Unattended call transfer - Auto answer - Call hold and resume - Microphone mute - Call waiting - Call recording - Peer to peer calls - Video calls - Instant Messaging - Custom ring tones - Repeat last call (redial) - Message Waiting Indication (MWI) - Address book with presence - Conferencing * Signaling: - SIP protocol support - SIP outbound protocol for NAT-traversal - SIP Re-invite - SIP Routes - SIP early media support - DNS NAPTR/SRV support - Multiple accounts support - DTMF support (RTP, SIP INFO) - Multicast sending & receiving * Security: - Signalling encryption (TLS) - Audio and video encryption (Secure RTP) - DTLS-SRTP key exchange protocol - ZRTP key exchange protocol - SDES key exchange protocol * Audio: - Low latency audio pipeline - High definition audio codecs - Audio device configuration - Audio filter plugins - Internal audio resampler for fixed sampling rates - Linear 16 bit wave format support for ringtones - Packet loss concealment (PLC) - Configurable ringtone playback device - Automatic gain control (AGC) and Noise reducation - Acoustic echo control (AEC) - Configurable audio sample format (Signed 16-bit, 24-bit, Float etc) - EBU ACIP (Audio Contribution over IP) Profile * Audio-codecs: - AAC - aptX - AMR narrowband, AMR wideband - Codec2 - G.711 - G.722 - G.726 - L16 - Opus * Audio-drivers: - Advanced Linux Sound Architecture (ALSA) audio-driver - PulseAudio POSIX OSes audio-driver - Android AAudio and OpenSLES audio-driver - Gstreamer playbin input audio-driver - JACK Audio Connection Kit audio-driver - MacOSX/iOS coreaudio/audiounit audio-driver - Portaudio audio-driver - Windows WASAPI audio-driver * Video: - Support for H.264, H.265, VP8, VP9, AV1 Video - Configurable resolution/framerate/bitrate - Configurable video input/output - Support for asymmetric video - Configurable video pixel format - Hardware acceleration for video encoder/decoder * Video-codecs: - AV1 - H.264 - H.265 - VP8 - VP9 * Video-drivers: - iOS avcapture video-source - FFmpeg/libav libavformat/avdevice input - Direct Show video-source - MacOSX AVCapture video-source - Linux V4L/V4L2 video-source - X11 grabber video-source - DirectFB video-output - SDL2 video-output - X11 video-output * NAT-traversal: - STUN support - TURN server support - ICE support - NATPMP support - PCP (Port Control Protocol) support * Networking: - multihoming, IPv4/IPv6 - automatic network roaming * Management: - Embedded web-server with HTTP interface - Command-line console over UDP/TCP - Command line interface (CLI) - Simple configuration files - MQTT (Message Queue Telemetry Transport) module * Profiles: - EBU ACIP (Audio Contribution over IP) Profile ## Building baresip is using CMake, and the following packages must be installed before building: * [libre](https://github.com/baresip/re) * [openssl](https://www.openssl.org/) See [Wiki: Install Stable Release](https://github.com/baresip/baresip/wiki/Install:-Stable-Release) or [Wiki: Install GIT Version](https://github.com/baresip/baresip/wiki/Install:-GIT-Version) for a full guide. ### Build with debug enabled ``` $ cmake -B build $ cmake --build build -j $ cmake --install build ``` ### Build with release ``` $ cmake -B build -DCMAKE_BUILD_TYPE=Release $ cmake --build build -j ``` ### Build with selected modules ``` $ cmake -B build -DMODULES="menu;account;g711" $ cmake --build build -j ``` ### Build with custom app modules ``` $ cmake -B build -DAPP_MODULES_DIR=../baresip-apps/modules -DAPP_MODULES="auloop;vidloop" $ cmake --build build -j ``` ### Build with clang compiler ``` $ cmake -B build -DCMAKE_C_COMPILER=clang -DCMAKE_CXX_COMPILER=clang++ $ cmake --build build -j ``` ### Build static ``` $ cmake -B build -DSTATIC=ON $ cmake --build build -j ``` Modules will be built if external dependencies are installed. After building you can start baresip like this: ``` $ build/baresip ``` The config files in `$HOME/.baresip` are automatically generated the first time you run baresip. ### Build documentation The API documentation can be build using [doxygen](https://www.doxygen.nl/manual/install.html). ``` $ doxygen mk/Doxyfile ``` By default the documentation is written to `../baresip-dox`, if you want to change the destination directory you can change the `OUTPUT_DIRECTORY` in `mk/Doxyfile`. ### Examples * Configuration examples are available in the [examples](https://github.com/baresip/baresip/tree/master/docs/examples) directory. * Documentation on configuring baresip can be found in the [Wiki](https://github.com/baresip/baresip/wiki/Configuration). ## License The baresip project is using the 3-clause BSD license. ## Contributing Patches can be sent via Github [Pull-Requests](https://github.com/baresip/baresip/pulls) or to the Baresip [mailing-list](https://groups.google.com/g/baresip). ## Design goals: * Minimalistic and modular VoIP client * SIP, SDP, RTP/RTCP, STUN/TURN/ICE * IPv4 and IPv6 support * RFC-compliancy * Robust, fast, low footprint * Portable C99 and C11 source code ## Modular Plugin Architecture: ``` aac Advanced Audio Coding (AAC) audio codec aaudio Android AAudio driver account Account loader alsa ALSA audio driver amr Adaptive Multi-Rate (AMR) audio codec aptx Audio Processing Technology codec (aptX) aubridge Audio bridge module auconv Audio sample format converter audiounit AudioUnit audio driver for MacOSX/iOS aufile Audio module for using a WAV-file as audio input augain Module to adjust gain of audio source auresamp Audio resampler ausine Audio sine wave input module av1 AV1 video codec avcapture Video source using iOS AVFoundation video capture avcodec Video codec using FFmpeg/libav libavcodec avfilter Video filter using FFmpeg libavfilter avformat Video source using FFmpeg/libav libavformat codec2 Codec2 low bit rate speech codec cons UDP/TCP console UI driver contact Contacts module coreaudio Apple macOS Coreaudio driver ctrl_dbus Control interface using DBUS ctrl_tcp TCP control interface using JSON payload debug_cmd Debug commands directfb DirectFB video display module dshow Windows DirectShow video source dtls_srtp DTLS-SRTP end-to-end encryption echo Echo server module evdev Linux input driver fakevideo Fake video input/output driver g711 G.711 audio codec g722 G.722 audio codec g7221 G.722.1 audio codec g726 G.726 audio codec gst Gstreamer audio source gtk GTK+ 3 menu-based UI gzrtp ZRTP module using GNU ZRTP C++ library httpd HTTP webserver UI-module httpreq HTTP request module ice ICE protocol for NAT Traversal in_band_dtmf In-band DTMF decoder jack JACK Audio Connection Kit audio-driver l16 L16 audio codec menu Interactive menu mixausrc Mixes another audio source into audio stream mixminus Mixes N-1 audio streams for conferencing mqtt MQTT (Message Queue Telemetry Transport) module mwi Message Waiting Indication natpmp NAT Port Mapping Protocol (NAT-PMP) module netroam Detects and applies changes of the local network addresses opensles OpenSLES audio driver opus OPUS Interactive audio codec opus_multistream OPUS multistream audio codec pcp Port Control Protocol (PCP) module plc Packet Loss Concealment (PLC) using spandsp portaudio Portaudio driver pulse Pulseaudio driver presence Presence module rtcpsummary RTCP summary module sdl Simple DirectMedia Layer 2.0 (SDL) video output driver selfview Video selfview module serreg Serial registration snapshot Save video-stream as PNG images sndfile Audio dumper using libsndfile sndio Audio driver for OpenBSD srtp Secure RTP encryption (SDES) using libre SRTP-stack stdio Standard input/output UI driver stun Session Traversal Utilities for NAT (STUN) module swscale Video scaling using libswscale syslog Syslog module turn Obtaining Relay Addresses from STUN (TURN) module uuid UUID generator and loader v4l2 Video4Linux2 video source vidbridge Video bridge module vidinfo Video info overlay module vp8 VP8 video codec vp9 VP9 video codec vumeter Display audio levels in console wasapi Windows Audio Session API (WASAPI) driver webrtc_aec Acoustic Echo Cancellation (AEC) using WebRTC SDK webrtc_aecm Acoustic Echo Cancellation (AEC) mobile using WebRTC SDK wincons Console input driver for Windows x11 X11 video output driver ``` ## IETF RFC/I-Ds: * RFC 3016 RTP Payload Format for MPEG-4 Audio/Visual Streams * RFC 3262 Reliability of Provisional Responses for SIP * RFC 3311 SIP UPDATE Method * RFC 3428 SIP Extension for Instant Messaging * RFC 3711 The Secure Real-time Transport Protocol (SRTP) * RFC 3640 RTP Payload Format for Transport of MPEG-4 Elementary Streams * RFC 3856 A Presence Event Package for SIP * RFC 3863 Presence Information Data Format (PIDF) * RFC 3891 The SIP "Replaces" Header * RFC 4145 TCP-Based Media Transport in SDP * RFC 4240 Basic Network Media Services with SIP (partly) * RFC 4347 Datagram Transport Layer Security * RFC 4568 SDP Security Descriptions for Media Streams * RFC 4572 Connection-Oriented Media Transport over TLS Protocol in SDP * RFC 4574 The SDP Label Attribute * RFC 4585 Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF) * RFC 4587 RTP Payload Format for H.261 Video Streams * RFC 4796 The SDP Content Attribute * RFC 4867 RTP Payload Format for the AMR and AMR-WB Audio Codecs * RFC 4961 Symmetric RTP / RTP Control Protocol (RTCP) * RFC 5285 A General Mechanism for RTP Header Extensions * RFC 5373 Requesting Answering Modes for SIP * RFC 5506 Support for Reduced-Size RTCP * RFC 5576 Source-Specific Media Attributes in SDP * RFC 5577 RTP Payload Format for ITU-T Recommendation G.722.1 * RFC 5626 Managing Client-Initiated Connections in SIP * RFC 5627 Obtaining and Using GRUUs in SIP * RFC 5761 Multiplexing RTP Data and Control Packets on a Single Port * RFC 5763 Framework for Establishing a SRTP Security Context Using DTLS * RFC 5764 DTLS Extension to Establish Keys for SRTP * RFC 5888 The SDP Grouping Framework * RFC 6157 IPv6 Transition in SIP * RFC 6184 RTP Payload Format for H.264 Video * RFC 6263 App. Mechanism for Keeping Alive NAT Associated with RTP / RTCP * RFC 6416 RTP Payload Format for MPEG-4 Audio/Visual Streams * RFC 6464 A RTP Header Extension for Client-to-Mixer Audio Level Indication * RFC 6716 Definition of the Opus Audio Codec * RFC 6886 NAT Port Mapping Protocol (NAT-PMP) * RFC 7064 URI Scheme for STUN Protocol * RFC 7065 TURN Uniform Resource Identifiers * RFC 7310 RTP Payload Format for Standard apt-X and Enhanced apt-X Codecs * RFC 7587 RTP Payload Format for the Opus Speech and Audio Codec * RFC 7741 RTP Payload Format for VP8 Video * RFC 7742 WebRTC Video Processing and Codec Requirements * RFC 7798 RTP Payload Format for High Efficiency Video Coding (HEVC) * RFC 8285 A General Mechanism for RTP Header Extensions * RFC 8843 Negotiating Media Multiplexing Using SDP * draft-ietf-payload-vp9-16 * RTP Payload Format For AV1 ## Supported platforms: * Android (8.0 or later) * Apple MacOS 11 and later (Xcode 10 or later) * Apple iOS 10.0 or later * Linux (kernel 4.0 or later, and glibc 2.31 or later) * Windows 10 or later (mingw and VS2022) ### Supported versions of C Standard library * Android bionic * BSD libc * GNU C Library (glibc) * Musl * Windows C Run-Time Libraries (CRT) * uClibc ### Supported compilers: * clang 9.x or later * gcc 9.x or later * MSVC 2022 or later ### Supported versions of OpenSSL * OpenSSL version 3.x.x * LibreSSL version 3.x ## Related projects * [libre - baresip fork](https://github.com/baresip/re) * [retest - baresip fork](https://github.com/baresip/retest) * [libre](https://github.com/creytiv/re) * [retest](https://github.com/creytiv/retest) ## References * Github: https://github.com/baresip/baresip * Mailing-list: https://groups.google.com/g/baresip