Baresip is a portable and modular SIP User-Agent with audio and video support Copyright (c) 2010 - 2013 Creytiv.com

Distributed under BSD license

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 C89 and C99 source code

Modular Plugin Architecture:

• alsa ALSA audio driver
• amr Adaptive Multi-Rate (AMR) audio codec
• audiounit AudioUnit audio driver for MacOSX/iOS
• auloop Audio-loop test module
• avcapture Video source using iOS AVFoundation video capture
• avcodec Video codec using FFmpeg
• avformat Video source using FFmpeg libavformat
• bv32 BroadVoice32 audio codec
• cairo Cairo video source
• celt CELT audio codec
• cons UDP console
• contact Contacts module
• coreaudio Apple Coreaudio driver
• evdev Linux input driver
• g711 G.711 audio codec
• g722 G.722 audio codec
• g7221 G.722.1 audio codec
• g726 G.726 audio codec
• gsm GSM audio codec
• gst Gstreamer audio source
• ice ICE protocol for NAT Traversal
• ilbc iLBC audio codec
• isac iSAC audio codec
• l16 L16 audio codec
• mda Symbian Mediaserver audio driver
• menu Interactive menu
• mwi Message Waiting Indication
• natbd NAT Behavior Discovery Module
• opengl OpenGL video output
• opengles OpenGLES video output
• opensles OpenSLES audio driver
• opus OPUS Interactive audio codec
• oss Open Sound System (OSS) audio driver
• plc Packet Loss Concealment (PLC) using spandsp
• portaudio Portaudio driver
• presence Presence module
• qtcapture Apple QTCapture video source driver
• quicktime Apple Quicktime video source driver
• rst Radio streamer using mpg123
• sdl Simple DirectMedia Layer (SDL) video output driver
• selfview Video selfview module
• silk SILK audio codec
• snapshot Save video-stream as PNG images
• sndfile Audio dumper using libsndfile
• speex Speex audio codec
• speex_aec Acoustic Echo Cancellation (AEC) using libspeexdsp
• speex_pp Audio pre-processor using libspeexdsp
• speex_resamp Speex resampler (deprecated)
• srtp Secure RTP encryption
• stdio Standard input/output UI driver
• stun Session Traversal Utilities for NAT (STUN) module
• syslog Syslog module
• turn Obtaining Relay Addresses from STUN (TURN) module
• uuid UUID generator and loader
• v4l Video4Linux video source
• v4l2 Video4Linux2 video source
• vidloop Video-loop test module
• vpx VP8/VPX video codec
• vumeter Display audio levels in console
• winwave Audio driver for Windows
• x11 X11 video output driver
• x11grab X11 grabber video source

IETF RFC/I-Ds:

• RFC 2190 RTP Payload Format for H.263 Video Streams (Historic)

• RFC 2429 RTP Payload Format for 1998 ver of ITU-T Rec. H.263 Video (H.263+)

• RFC 3016 RTP Payload Format for MPEG-4 Audio/Visual Streams

• RFC 3711 The Secure Real-time Transport Protocol (SRTP)

• RFC 3856 A Presence Event Package for SIP

• RFC 3863 Presence Information Data Format (PIDF)

• RFC 3951 Internet Low Bit Rate Codec (iLBC)

• RFC 3952 RTP Payload Format for iLBC Speech

• RFC 3984 RTP Payload Format for H.264 Video

• RFC 4240 Basic Network Media Services with SIP (partly)

• RFC 4298 Broadvoice Speech Codecs

• RFC 4568 SDP Security Descriptions for Media Streams

• 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 4629 RTP Payload Format for ITU-T Rec. H.263 Video

• 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 5168 XML Schema for Media Control

• RFC 5574 RTP Payload Format for the Speex Codec

• RFC 5577 RTP Payload Format for ITU-T Recommendation G.722.1

• RFC 5626 Managing Client-Initiated Connections in SIP

• RFC 5761 Multiplexing RTP Data and Control Packets on a Single Port

• RFC 6263 App. Mechanism for Keeping Alive NAT Associated with RTP / RTCP

• RFC 6716 Definition of the Opus Audio Codec

• draft-valin-celt-rtp-profile-02

• draft-ietf-payload-vp8-08

• draft-spittka-payload-rtp-opus-00

Architecture:

                   .------.
                   |Video |
                 _ |Stream|\
                 /|'------' \ 1
                /            \
               /             _\|
 .--. N  .----. M  .------. 1  .-------. 1  .-----.
 |UA|--->|Call|--->|Audio |--->|Generic|--->|Media|
 '--'    '----'    |Stream|    |Stream |    | NAT |
            |1     '------'    '-------'    '-----'
            |         C|       1|   |
           \|/      .-----.  .----. |
        .-------.   |Codec|  |Jbuf| |1
        | SIP   |   '-----'  '----' |
        |Session|     1|       /|\  |
        '-------'    .---.      |  \|/
                     |DSP|    .--------.
                     '---'    |RTP/RTCP|
                              '--------'
                              |  SRTP  |
                              '--------'

   A User-Agent (UA) has 0-N SIP Calls
   A SIP Call has 0-M Media Streams

Supported platforms:

• Linux
• FreeBSD
• OpenBSD
• NetBSD
• Symbian OS
• Solaris
• Windows
• Apple Mac OS X and iOS
• Android

Supported compilers:

• gcc (v2.9x to v4.x)
• gcce
• ms vc2003 compiler
• codewarrior

External dependencies:

libre librem

Feedback:

• Please send feedback to libre@creytiv.com