开源sip server & sip client 和开发库 一览
http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software
http://blog.csdn.net/xuyunzhang/article/details/26859341
Asterisk
Asterisk is an open source framework for building communications applications. Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and other custom solutions. It is used by small businesses, large businesses, call centers, carriers and government agencies, worldwide. Asterisk is free and open source. Asterisk is sponsored by Digium.
SIP Proxies
- Mini-SIP-Proxy A very tiny perl POE based SIP proxy
- MjServer cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack
- MySIPSwitch SIP Proxy server which allows using multiple SIP accounts with a single SIP login
- NethidPro3.0.6 Opensource Sip Encryption Bridge: www.vonets.com
- Net-SIP A Perl SIP framework that includes a stateless proxy
- OpenJSIP Opensource distributed standalone SIP proxy, SIP registrar, SIP location service run by Java VM. Based on OpenSIPS forked from OpenSER.
- partysip SIP proxy server
- SaRP SIP and RTP Proxy in Perl
- sipd SIP Proxy
- theSIP router/proxy/jack-in-all-trades from IPtel.org
- Siproxd SIP and RTP Proxy
- SIPVicious tool suite: tools for auditing sip devices
- sipX The SIP PBX for Linux: Complete, native SIP PBX solution from Vocal SIP softswitch with H.323 and MGCP translators for non-SIP endpoints
- Yxa Written in the Erlang programming language
SIP Clients (UA's)
Linux clients:
- H.323 audio
and video softphone for various linux, solaris, windows, and various unix systems. Formerly Linphone audio
and video SIP softphone for Linux and Windows XP - minisip cross-platform
SIP softphone, Linux, Windows XP and soon Windows Mobile 2003 SE - OpenSIPStack MPL
licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal.
Reference implementation of Session Border Controller (OpenSBC)
available. - Peers Minimalist
SIP softphone written in java (tested on linux and windows) - SIP softphone
in Python, runs on Windows, Mac, Linux - SipToSis frommhspot.com Skype
SIP UA - Multiplatform - Open Source - YateClient is
multiprotocol and multiplatform softphone with H.323, SIP, Jingle and IAX support. - Linphone backend
MacOS X clients:
- SIP softphone
in Python, runs on Windows, Mac, Linux - http://www.mhspot.com Skype
SIP UA - Multiplatform - Open Source - YateClient skinnable
VoIP client based on QT library which supports H.323, SIP, Jingle and IAX protocols
Windows clients
- H.323 audio
and video softphone for various linux, solaris, windows, and various unix systems. Formerly JPhone Rich
software SDK support softphone development, Windows, Linux, ThreadX, Vxworks etc. - Linphone audio
and video SIP softphone for Linux and Windows XP - minisip cross-platform
SIP softphone, Linux, Windows XP and soon Windows Mobile 2003 SE - OpenSIPStack MPL
licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal.
Reference implementation of Session Border Controller (OpenSBC)
available. - Peers Minimalist
SIP softphone written in java (tested on linux and windows) - SIP softphone
in Python, runs on Windows, Mac, Linux - SipToSis frommhspot.com Skype
SIP UA - Multiplatform - Open Source - wxCommunicator Windows
softphone based on sipXtapi and wxWidgets 2.8.x, multi-account, conferencing, NAT support - YateClient is
multiprotocol and multiplatform softphone with H.323, SIP,Jingle and IAX support.
SIP tools
- Open Source Asterisk AMI: Open Source Asterisk AMI interface application
- SIP
SIMPLE Command Line Tools for SIP sessions (complete console
based SIP UA) and SIMPLE Presence (Publish, Subscribe, Notify) and XCAP
document manipulation SIP Protocol Stacks and Libraries
- Aloha Spring
based J2SE SIP A/S which leverages optimistic concurrent model and supports multiple persistence models - eXosip -
eXtended osip library - libdissipate SIP
stack - minisip includes
a SIP stack - MjSip -
complete and powerful java-based SIP library for both J2SE and J2ME platforms. - Open
Sip Stack MPL licensed SIP stack with ENUM, Presence
(XMPP/SIMPLE) and NAT traversal. Reference implementation of Session
Border Controller (OpenSBC) available. - Verona based
Active/X plugin for IE allowing ClickToDial functionallity - reSIProcate SIP
stack and sample Application from SailFin Adds
SIP support the the Java GlassFish Application Server - sipXtackLib an
RFC 3261, 3263 complient SIP stack from http://sofia-sip.sourceforge.net Sofia-Sip
is SIP stack implementation with STUN and presense support - Twisted Python
protocol stacks and applications includes SIP support - Verona -
GPL licenesed VOIP engine based on oSIP,eXosip,oRTP,ffmepg, works on Linux,Windows Mac-OS/X - YASS - Statefull SIP stack used inYate written
in C++ usable for client, server or proxy in a multithread or single
thread model. It's working on both Windows and Linux, it's very small
but full featured.
H.323 Clients
Linux clients:
- H.323 audio
and video softphone for various linux, solaris, windows, and various unix systems. Formerly YateClient is
multiprotocol and multiplatform softphone with H.323, SIP and IAX support.
MacOS X clients:
- YateClient skinnable
VoIP client based on the QT library which supports H.323, SIP, Jingle and IAX protocols
Windows clients:
- H.323 audio
and video softphone for various linux, solaris, windows, and various unix systems. Formerly YateClient is
multiprotocol and multiplatform softphone with H.323, SIP and IAX support.
H.323 Gatekeeper
- GNU
Gatekeeper - for Linux, Windows, Mac etc.
IAX clients
- IAXComm for
Linux, MacOS X and Windows - Kiax -
for Linux, Windows and MacOS, based on iaxclient, GPL - QtIax fromYateClient is
multiprotocol and multiplatform softphone with H.323, SIP and IAX support.
RTP Proxies
RTP Protocol Stacks
- ccRTP C++
library based on GNU Common C++ - JRTPLIB C++
object oriented RTP library - libRTP part
of gnome-o-phone - libzrtpcpp -
ZRTP extension library for ccRTP stack - oRTP Written
in C, running on linux, win32 and arm-linux. - RTPlib C
library - sipXmediaLib RTP
+ audio bridges, audio splitters, echo suppression, tone from
generation (e.g. DTMF), streaming support, RTCP, G711 codecs, etc. from Secure RTP - see:YRTP -Yate RTP
stack, that can be used in other projects. - zrtp4j -
ZRTP stack for Java, based on GNU ZRTP, used in SIP Communicator
MSRP Relays
XCAP servers
Other tools
- Encours Teleconferencing
in your web browser with an integrated VOIP layer (Java) and an optional Asterisk connectivity on the server side. - MORCC -
automated online Calling Card store. Paypal integrated. - OgonPhonesXML .NET
Library for Aastra SIP Phones and Cisco SIP/IP phones for fast and easy XML Interfacement. - Oreka capture
and retrieval of SIP, Cisco Skinny (SCCP) and raw RTP sessions with
audio compression, rdbms metadata storage and web based user interface. - Voipong -
Voice over IP (VoIP) sniffer and call detector. - Vomit converts
a Cisco IP phone conversation (recorded with TCPdump) into a standard WAV file
PBX platforms
Some of these include SIP proxy functionality
- H.323 and
other protocols - FreeSWITCH Open
Source PBX and Soft Switch - sipX -
The SIP PBX for Linux from YATE Yet
Another Telephony Engine - supportsIVR platforms
- YATE Yet
Another Telephony Engine - See Also:
Voicemail servers
- YATE Yet
Another Telephony Engine with H.323, SIP and IAX support.
Speech
Text-to-speech and speech-to-text (voice recognition)
Fax Servers
Development platforms, protocol stacks
- H.323 Protocol
Stack following on from the original openH323 - SS7 Protocol
Stack - H.323 Protocol
Stack Developed in C - ++Skype C++
library for skype add-onplatform independent software development. It is platform independent,
easy to use, and easy to extend because of the flexible library design,
inspired by modern C++ design ideas. Performance is one of the goals.
Radius Servers
- RadBox RADIUS
Server + Billing System. (For a work, you nead instal Framework 2.0)
Billing
- See
Codecs
- See
Middleware
Suite Solutions
CTI Dialer utilities
- TALK Powerful
directory management and scalable architecture to create Click to call
or Select and Dial applications + AJAX libraries to implement these
features in your web site.
- TALK Powerful
- See
- H.323 Protocol
- YATE Yet
- YATE Yet
- Encours Teleconferencing
- ccRTP C++
- Aloha Spring
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