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https://github.com/baresip/baresip
Baresip is a modular SIP User-Agent with audio and video support
https://github.com/baresip/baresip
libre sip telephony voip webrtc
Last synced: 14 days ago
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Baresip is a modular SIP User-Agent with audio and video support
- Host: GitHub
- URL: https://github.com/baresip/baresip
- Owner: baresip
- License: bsd-3-clause
- Created: 2014-02-09T10:44:09.000Z (over 10 years ago)
- Default Branch: main
- Last Pushed: 2024-05-21T09:01:10.000Z (about 1 month ago)
- Last Synced: 2024-05-21T11:15:50.220Z (about 1 month ago)
- Topics: libre, sip, telephony, voip, webrtc
- Language: C
- Homepage:
- Size: 8.1 MB
- Stars: 1,630
- Watchers: 90
- Forks: 427
- Open Issues: 10
-
Metadata Files:
- Readme: README.md
- Changelog: CHANGELOG.md
- Contributing: docs/CONTRIBUTING.md
- License: LICENSE
Lists
- docsify-awesome-stars - baresip/baresip - Baresip is a modular SIP User-Agent with audio and video support (C)
- awesome-telephony - baresip - Agent with audio and video support (SIP Clients)
- awesome-stars - baresip - Agent with audio and video support | baresip | 813 | (C)
- awesome-voip - Baresip
- awesome-starred - baresip/baresip - Baresip is a modular SIP User-Agent with audio and video support (others)
README
baresip README
==============
Baresip is a portable and modular SIP User-Agent with audio and video support.
Copyright (c) 2010 - 2024 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)
- EBU ACIP (Audio Contribution over IP) Profile
* Audio-codecs:
- AAC
- aptX
- AMR narrowband, AMR wideband
- Codec2
- G.711
- G.722
- G.726
- L16
- MPA
- Opus
* Audio-drivers:
- Advanced Linux Sound Architecture (ALSA) audio-driver
- PulseAudio POSIX OSes audio-driver
- Android OpenSLES audio-driver
- Gstreamer playbin input audio-driver
- JACK Audio Connection Kit audio-driver
- MacOSX/iOS coreaudio/audiounit audio-driver
- Portaudio audio-driver
- Windows winwave 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
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
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
ebuacip EBU ACIP (Audio Contribution over IP) Profile
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
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
mpa MPA Speech and Audio Codec
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
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
winwave Audio driver for Windows
x11 X11 video output driver
```
## IETF RFC/I-Ds:
* RFC 2250 RTP Payload Format for the mpa Speech and Audio Codec
* 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
## Supported platforms:
* Android (6.0 or later)
* Apple macOS (10.12+)
* Apple iOS 9.0 or later
* Linux (kernel 3.0 or later, and glibc 2.5.x or later)
* OpenBSD
* Windows 10 or later (mingw and VS2019)
### Supported versions of C Standard library
* Android bionic
* BSD libc
* GNU C Library (glibc)
* Musl
* Windows C Run-Time Libraries (CRT)
* uClibc
### Supported compilers:
* gcc 4.9 or later
* MSVC 2019
* clang 3.x or later
### Supported versions of OpenSSL
* OpenSSL version 1.1.1
* 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