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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

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

      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 - 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.

05-07 15:43
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