baresip-0.4.7
所属分类:IP电话/视频会议
开发工具:Unix_Linux
文件大小:407KB
下载次数:17
上传日期:2013-11-27 19:09:00
上 传 者:
netpanther
说明: 一个小巧,性能不错的sip协议栈,c语言编写:
* 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
(* 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)
文件列表:
baresip-0.4.7\docs\ChangeLog (12316, 2013-11-13)
baresip-0.4.7\docs\TODO (969, 2013-09-04)
baresip-0.4.7\docs\COPYING (1539, 2013-01-01)
baresip-0.4.7\share\message.wav (16684, 2010-11-03)
baresip-0.4.7\share\busy.wav (18459, 2010-11-03)
baresip-0.4.7\share\ringback.wav (68084, 2010-11-03)
baresip-0.4.7\share\error.wav (21463, 2010-11-03)
baresip-0.4.7\share\ring.wav (35184, 2010-11-03)
baresip-0.4.7\share\notfound.wav (15760, 2010-11-03)
baresip-0.4.7\share\callwaiting.wav (80418, 2012-05-15)
baresip-0.4.7\src\rtpkeep.c (3082, 2013-06-16)
baresip-0.4.7\src\config.c (17125, 2013-11-13)
baresip-0.4.7\src\sipreq.c (2437, 2013-10-02)
baresip-0.4.7\src\ui.c (2944, 2013-10-01)
baresip-0.4.7\src\audio.c (27917, 2013-10-25)
baresip-0.4.7\src\core.h (9573, 2013-11-13)
baresip-0.4.7\src\call.c (32333, 2013-10-12)
baresip-0.4.7\src\vidisp.c (2533, 2013-10-01)
baresip-0.4.7\src\realtime.c (2144, 2013-11-13)
baresip-0.4.7\src\magic.h (558, 2010-11-03)
baresip-0.4.7\src\mctrl.c (763, 2010-11-03)
baresip-0.4.7\src\contact.c (2731, 2012-08-26)
baresip-0.4.7\src\reg.c (5424, 2013-09-25)
baresip-0.4.7\src\mnat.c (1565, 2011-11-07)
baresip-0.4.7\src\vidfilt.c (1686, 2013-09-02)
baresip-0.4.7\src\cmd.c (5312, 2013-10-13)
baresip-0.4.7\src\aufilt.c (445, 2013-01-03)
baresip-0.4.7\src\srcs.mk (690, 2013-08-03)
baresip-0.4.7\src\net.c (7542, 2013-10-25)
baresip-0.4.7\src\play.c (5043, 2013-06-19)
baresip-0.4.7\src\ua.c (28383, 2013-11-13)
baresip-0.4.7\src\ausrc.c (1995, 2012-03-31)
baresip-0.4.7\src\module.c (2533, 2013-10-19)
baresip-0.4.7\src\message.c (2432, 2013-07-04)
baresip-0.4.7\src\menc.c (996, 2013-01-19)
baresip-0.4.7\src\vidcodec.c (1186, 2013-01-11)
baresip-0.4.7\src\auplay.c (1945, 2012-03-31)
baresip-0.4.7\src\conf.c (6423, 2013-11-13)
baresip-0.4.7\src\sdp.c (3307, 2013-08-11)
... ...
README
------
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:
account Account loader
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
directfb DirectFB video display module
dshow Windows DirectShow video source
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
httpd HTTP webserver UI-module
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
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
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
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
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
wincons Console input driver for Windows
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 19*** 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 3***4 RTP Payload Format for H.2*** Video
* RFC 4240 Basic Network Media Services with SIP (partly)
* RFC 42*** 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 5506 Support for Reduced-Size RTCP
* 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 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)
* 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
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