开源sip server & sip client 和开发库 一览

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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
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  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

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    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

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      XCAP servers

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        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)

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              Fax Servers

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                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

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                      Suite Solutions

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                        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.

开源sip server & sip client 和开发库 一览

标签：

原文地址：http://www.cnblogs.com/welhzh/p/4601683.html