README: change to markdown format

2 files changed, 415 insertions, 347 deletions

diff --git a/README.md b/README.md
index e59aaf6..22eccd8 100644
--- a/README.md
+++ b/README.md
@@ -1,18 +1,424 @@
-baresip
-=======
+baresip 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
+
 
 [![Build Status](https://travis-ci.org/alfredh/baresip.svg?branch=master)](https://travis-ci.org/alfredh/baresip)
 
-Baresip is a modular SIP User-Agent with audio and video support
 
-License: BSD
+## 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
+
+
+## Building
+
+baresip is using GNU makefiles, and the following packages must be
+installed before building:
+
+* [libre](https://github.com/creytiv/re)
+* [librem](https://github.com/creytiv/rem)
+* [openssl](https://www.openssl.org/)
+
+
+### Build with debug enabled
+
+```$ make
+$ sudo make install
+$ sudo ldconfig```
+
+### Build with release
+
+```
+$ make RELEASE=1
+$ sudo make RELEASE=1 install
+$ sudo ldconfig
+```
+
+### Build with clang compiler
+
+```
+$ make CC=clang
+$ sudo make CC=clang install
+$ sudo ldconfig
+```
+
+
+## Documentation
+
+The online documentation generated with doxygen is available in
+the main [website](http://creytiv.com/doxygen/baresip-dox/html/)
+
+
+### Examples
+
+Configuration examples are available from the
+[examples](https://github.com/alfredh/baresip/tree/master/docs/examples)
+directory.
+
+
+## License
+
+The baresip project is using the BSD license.
+
+
+## Contributing
+
+Patches can sent via Github
+[Pull-Requests](https://github.com/creytiv/baresip/pulls) or to the RE devel
+[mailing-list](http://lists.creytiv.com/mailman/listinfo/re-devel).
+
+
+## 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:
+(note: out of date, needs updating)
+
+```
+                   .------.
+                   |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:
+
+* Android
+* Apple Mac OS X and iOS
+* FreeBSD
+* Linux
+* NetBSD
+* OpenBSD
+* Solaris
+* Windows
+
+
+### Supported versions of C Standard library
+
+* Android bionic
+* BSD libc
+* GNU C Library (glibc)
+* Windows C Run-Time Libraries (CRT)
+* uClibc
+
+
+### Supported compilers:
+
+* gcc 3.x
+* gcc 4.x
+* gcc 5.x
+* gcc 6.x
+* ms vc2003 compiler
+* clang
+
+### Supported versions of OpenSSL
+
+* OpenSSL version 1.0.1
+* OpenSSL version 1.0.2
+* OpenSSL version 1.1.0
+* LibreSSL version 2.x
+
 
+## Related projects
 
-see [README](docs/README) for more details
+* [libre](https://github.com/creytiv/re)
+* [librem](https://github.com/creytiv/rem)
+* [retest](https://github.com/creytiv/retest)
+* [restund](http://creytiv.com/restund.html)
 
 
-# Resources
+## References
 
-- Project homepage: http://www.creytiv.com/baresip.html
-- Github: https://github.com/alfredh/baresip
-- Mailing-list: http://lists.creytiv.com/mailman/listinfo/re-devel
+* Project homepage: http://www.creytiv.com/baresip.html
+* Github: https://github.com/alfredh/baresip
+* Mailing-list: http://lists.creytiv.com/mailman/listinfo/re-devel
diff --git a/docs/README b/docs/README
deleted file mode 100644
index 5f60ab5..0000000
--- a/docs/README
+++ /dev/null
@@ -1,338 +0,0 @@
-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>