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-README
-------
-
-Baresip is a portable and modular SIP User-Agent with audio and video support
-Copyright (c) 2010 - 2016 Creytiv.com
-
-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
-
-* 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)
-
-* 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)
-
-* Audio-codecs:
- - AMR narrowband, AMR wideband
- - BroadVoice32 BV32
- - Codec2
- - G.711
- - G.722
- - G.726
- - GSM
- - iLBC
- - iSAC
- - L16
- - MPA
- - Opus
- - Silk
- - Speex
-
-* Audio-drivers:
- - Advanced Linux Sound Architecture (ALSA) audio-driver
- - Android OpenSLES audio-driver
- - Gstreamer playbin input audio-driver
- - JACK Audio Connection Kit audio-driver
- - MacOSX/iOS coreaudio/audiounit audio-driver
- - Open Sound System (OSS) audio-driver
- - Portaudio audio-driver
- - Windows winwave audio-driver
-
-* Video:
- - Support for H.265, H.264, H.263, VP8, VP9, MPEG-4 Video
- - Configurable resolution/framerate/bitrate
- - Configurable video input/output
- - Support for asymmetric video
-
-* Video-codecs:
- - H.265
- - H.264
- - H.263
- - VP8
- - VP9
- - MPEG-4
-
-* Video-drivers:
- - iOS avcapture video-source
- - FFmpeg/libav libavformat/avdevice input
- - Cairo video-source test module
- - Direct Show video-source
- - MacOSX QTcapture/quicktime video-source
- - RST media player
- - Linux V4L/V4L2 video-source
- - X11 grabber video-source
- - DirectFB video-output
- - OpenGL/OpenGLES video-output
- - SDL/SDL2 video-output
- - X11 video-output
-
-* NAT-traversal:
- - STUN support
- - TURN server support
- - ICE and ICE-lite support
- - NATPMP 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
-
-
-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:
-
-account Account loader
-alsa ALSA audio driver
-amr Adaptive Multi-Rate (AMR) audio codec
-aubridge Audio bridge module
-audiounit AudioUnit audio driver for MacOSX/iOS
-aufile Audio module for using a WAV-file as audio input
-auloop Audio-loop test module
-avcapture Video source using iOS AVFoundation video capture
-avcodec Video codec using FFmpeg/libav libavcodec
-avformat Video source using FFmpeg/libav libavformat
-b2bua Back-to-Back User-Agent (B2BUA) module
-bv32 BroadVoice32 audio codec
-cairo Cairo video source
-codec2 Codec2 low bit rate speech codec
-cons UDP/TCP console UI driver
-contact Contacts module
-coreaudio Apple Coreaudio driver
-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
-gsm GSM audio codec
-gst Gstreamer audio source
-gst1 Gstreamer 1.0 audio source
-gst_video Gstreamer video codec
-gst_video1 Gstreamer 1.0 video codec
-gtk GTK+ 2.0 UI
-h265 H.265 video codec
-httpd HTTP webserver UI-module
-ice ICE protocol for NAT Traversal
-ilbc iLBC audio codec
-isac iSAC audio codec
-jack JACK Audio Connection Kit audio-driver
-l16 L16 audio codec
-libsrtp Secure RTP encryption using libsrtp
-menu Interactive menu
-mpa MPA Speech and Audio Codec
-mwi Message Waiting Indication
-natbd NAT Behavior Discovery Module
-natpmp NAT Port Mapping Protocol (NAT-PMP) 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
-pcp Port Control Protocol (PCP) module
-plc Packet Loss Concealment (PLC) using spandsp
-portaudio Portaudio driver
-pulse Pulseaudio driver
-presence Presence module
-qtcapture Apple QTCapture video source driver
-quicktime Apple Quicktime video source driver (deprecated)
-rst Radio streamer using mpg123
-sdl Simple DirectMedia Layer (SDL) video output driver
-sdl2 Simple DirectMedia Layer v2 (SDL2) video output driver
-selfview Video selfview module
-silk SILK audio codec
-snapshot Save video-stream as PNG images
-sndfile Audio dumper using libsndfile
-sndio Audio driver for OpenBSD
-speex Speex audio codec
-speex_aec Acoustic Echo Cancellation (AEC) using libspeexdsp
-speex_pp Audio pre-processor using libspeexdsp
-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
-v4l Video4Linux video source
-v4l2 Video4Linux2 video source
-v4l2_codec Video4Linux2 video codec module (H264 hardware encoding)
-vidbridge Video bridge module
-vidinfo Video info overlay module
-vidloop Video-loop test module
-vp8 VP8 video codec
-vp9 VP9 video codec
-vumeter Display audio levels in console
-wincons Console input driver for Windows
-winwave Audio driver for Windows
-x11 X11 video output driver
-x11grab X11 grabber video source
-zrtp ZRTP media encryption module
-
-
-IETF RFC/I-Ds:
-
-* RFC 2190 RTP Payload Format for H.263 Video Streams (Historic)
-* RFC 2250 RTP Payload Format for the mpa Speech and Audio Codec
-* 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 3428 SIP Extension for Instant Messaging
-* 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 4145 TCP-Based Media Transport in SDP
-* RFC 4240 Basic Network Media Services with SIP (partly)
-* RFC 4298 Broadvoice Speech Codecs
-* 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 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 5506 Support for Reduced-Size RTCP
-* RFC 5574 RTP Payload Format for the Speex Codec
-* 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 5780 NAT Behaviour Discovery Using STUN
-* RFC 6263 App. Mechanism for Keeping Alive NAT Associated with RTP / RTCP
-* RFC 6716 Definition of the Opus Audio Codec
-* RFC 6886 NAT Port Mapping Protocol (NAT-PMP)
-* RFC 7587 RTP Payload Format for the Opus Speech and Audio Codec
-* RFC 7741 RTP Payload Format for VP8 Video
-* RFC 7798 RTP Payload Format for High Efficiency Video Coding (HEVC)
-
-* draft-ietf-avt-rtp-isac-04
-
-
-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
-* Solaris
-* Windows
-* Apple Mac OS X and iOS
-* Android
-
-
-Supported compilers:
-
-* gcc (v2.9x to v4.x)
-* gcce
-* llvm clang
-* ms vc2003 compiler
-
-
-External dependencies:
-
-libre v0.4.14 or later
-librem v0.4.7 or later
-
-
-Feedback:
-
-- Please send feedback to <libre@creytiv.com>