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@@ -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 + [](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> |