baresip

baresip README

Baresip is a portable and modular SIP User-Agent with audio and video support.
Copyright (c) 2010 - 2026 Alfred E. Heggestad and Contributors
Distributed under BSD license

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)

• Audio-codecs:

  • AAC
  • aptX
  • AMR narrowband, AMR wideband
  • Codec2
  • G.711
  • G.722
  • 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
  • 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

Building

baresip is using CMake, and the following packages must be installed before building:

• libre
• openssl

See Wiki: Install Stable Release or 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.

$ 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 directory.
• Documentation on configuring baresip can be found in the Wiki.

License

The baresip project is using the 3-clause BSD license.

Contributing

Patches can be sent via Github Pull-Requests or to the Baresip mailing-list.

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:

…

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 4585 Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF)

• 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
• retest - baresip fork
• libre
• retest

References

• Github: https://github.com/baresip/baresip
• Mailing-list: https://groups.google.com/g/baresip