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

the SIP 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 from mhspot.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 from mhspot.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 in Yate 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 from YateClient 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 - supports
        IVR 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-on platform 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.