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开源sip server & sip client 和开发库 一览

简介: en source means all source code is available!! Do not post any "free but not open" software here! SIP Proxies Mini-SIP-Proxy A very tiny perl POE...
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en source means all source code is available!! Do not post any "free but not open" software here!

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
  • JAIN-SIP Proxy
  • OpenJSIP Opensource distributed standalone SIP proxy, SIP registrar, SIP location service run by Java VM. Based on NIST SIP and derived from JAIN-SIP Proxy.
  • OpenSBC: MPL licensed SIP proxy/registrar/B2BUA with NAT traversal and ENUM
  • OpenSER: GPL SIP Server with TLS support - renamed to Kamailio
  • OpenSIPS forked from OpenSER.
  • partysip SIP proxy server
  • SaRP SIP and RTP Proxy in Perl
  • sipd SIP Proxy
  • SIP Express Router (SER): 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 SIPfoundry
  • 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:

  • Cockatoo
  • Ekiga || SIPH.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
  • FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
  • Kphone
  • 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
  • MjUA: simple cross-platform SIP softphone, written in java, based on MjSip stack
  • Open IP Phone Business IP Phone sdk support, ims compliant, good interoperability.
  • OpenSIPStack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
  • OpenSoftphone: A simple Java based SIP softphone using the PjSip-jni wrapper.
  • OpenZoep: GPL telephone and IM messaging client engine
  • Peers Minimalist SIP softphone written in java (tested on linux and windows)
  • PhoneGaim
  • PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
  • QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
  • SFLphone, open-source multiplatform multi-protocol VoIP client
  • ShtoomSIP softphone in Python, runs on Windows, Mac, Linux
  • SIP Communicator Audio/Video phone and messenger - Multiplatform - Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
  • SipToSis from mhspot.com Skype SIP UA - Multiplatform - Open Source
  • sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
  • sipXphone from SIPfoundry, previously known as the Pingtel phone
  • Twinkle
  • YateClient is multiprotocol and multiplatform softphone with H.323, SIP, Jingle and IAX support.
  • YeaPhone: A SIP softphone for the Yealink USB-P1K handset based on the libLinphone backend

 

MacOS X clients:

  • Blink: It supports wideband VoIP, Instant Messaging, File Transfer and Desktop Sharing based on MSRP
  • FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
  • PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
  • QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
  • SFLphone, open-source multiplatform multi-protocol VoIP client
  • ShtoomSIP softphone in Python, runs on Windows, Mac, Linux
  • SIP Communicator Audio/Video phone and messenger - Multiplatform - Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
  • SipToSis from http://www.mhspot.com Skype SIP UA - Multiplatform - Open Source
  • Telephone: A SIP softphone designed for the Mac (written in Objective-C/Cocoa). Very good integration with Mac OSX : Dial from Addressbook, dial tel: URIs from Safari, notifications with Growl.
  • YateClient skinnable VoIP client based on QT library which supports H.323, SIP, Jingle and IAX protocols

 

Windows clients

  • Ekiga || SIPH.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
  • FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
  • 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
  • MjUA: simple cross-platform SIP softphone, written in java, based on MjSip stack
  • OfficeSIP Messenger is audio-video softphone and instant messenger, open source alternative to MS Office Communicator.
  • OfficeSIP Softphone GPL audio-video softphone.
  • OpenSIPStack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
  • OpenZoep: GPL telephone and IM messaging client engine
  • Peers Minimalist SIP softphone written in java (tested on linux and windows)
  • PhoneGaim
  • PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
  • OpenSoftphone: A simple Java based SIP softphone using the PjSip-jni wrapper
  • QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
  • ShtoomSIP softphone in Python, runs on Windows, Mac, Linux
  • SIP Communicator Audio/Video phone and messenger - Multiplatform - Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
  • SipToSis from mhspot.com Skype SIP UA - Multiplatform - Open Source
  • sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
  • sipXphone from SIPfoundry, previously known as the Pingtel phone
  • VMukti (formerly 1videoConference) alpha: a web2.0 VoIP video conferencing software for Asterisk.
  • 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

  • Callflow: Generates SIP Call Flow diagrams
  • miTester for SIP: SIP testing tool; Automates test execution.
  • Open Source Asterisk AMI: Open Source Asterisk AMI interface application
  • pjsip-perf: SIP transaction and call performance measurement tool
  • PROTOS Test-Suite: SIP Testing tools
  • SFTF: SIP Forum Test Framework - a SIP UA test suite primarily targeted at UA software developers hosted by SIPfoundry
  • SIP-CallerID: SIP Caller ID retrieval and lookup
  • SIPbomber: SIP proxy testing tool
  • SIP SIMPLE Command Line Tools for SIP sessions (complete console based SIP UA) and SIMPLE Presence (Publish, Subscribe, Notify) and XCAP document manipulation
  • Sipp: SIP performance tester
  • Sipper: SIPr (called Sipper) is an open source and a comprehensive SIP application testing framework. Generate any call flow in minutes.
  • SIP Proxy: SIP security testing tool.
  • Sipsak: SIP testing tool
  • SIP Soft client: Software development kit for SIP Softphone
  • SIPVicious tool suite: tools for auditing SIP devices
  • SMAP: Locating and fingerprinting remote SIP devices
  • Vovida.org load balancer: SIP Load Balancer

 

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
  • Juphoon SIP Stack Rich software SDK support SIP, SDP, XML, RTP/RTCP, HTTP, STUN, ABNF etc. Support Windows, Linux, ThreadX, Vxworks etc.
  • libdissipate SIP stack
  • minisip includes a SIP stack
  • MjSip - complete and powerful java-based SIP library for both J2SE and J2ME platforms.
  • MSRP Library - MSRP protocol (RFC4975) and its relay extension (RFC4976) written in Python
  • NIST SIP Various SIP appications and tools in Java
  • Open Sip Stack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
  • oSIP Library SIP Library
  • OSP client protocol stack and SIPfoundry
  • PhClickDial - Verona based Active/X plugin for IE allowing ClickToDial functionallity
  • PJSIP: Small footprint, high performance, and ultra-portable SIP stack written in C, and has language binding for Python. Works on smartphones (Symbian, Windows, iPhone/iOS, Android) as well as desktops and support ZRTP encryption.
  • reSIProcate SIP stack and sample Application from SIPfoundry
  • SailFin Adds SIP support the the Java GlassFish Application Server
  • sipXtackLib an RFC 3261, 3263 complient SIP stack from SIPfoundry
  • http://sofia-sip.sourceforge.net Sofia-Sip is SIP stack implementation with STUN and presense support
  • SIP SIMPLE client SDK - High level middleware on top of SIP, RTP, MSRP and XCAP protocols
  • 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
  • Vovida SIP Vovida SIP stack
  • XCAP Library - XCAP client library written in Python
  • 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:

 

MacOS X clients:

  • FreeSWITCH: Console client using OPAL
  • ohphoneX
  • YateClient skinnable VoIP client based on the QT library which supports H.323, SIP, Jingle and IAX protocols

 

Windows clients:

  • Ekiga || SIPH.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
  • FreeSWITCH: Console client using OPAL
  • OpenPhone
  • YateClient is multiprotocol and multiplatform softphone with H.323, SIP and IAX support.

 

H.323 Gatekeeper

 

IAX clients

 

RTP Proxies

 

RTP Protocol Stacks

  • ccRTP C++ library based on GNU Common C++
  • Juphoon RTP Stack Rich software SDK include RTP/RTCP stack. Support Windows, Linux, ThreadX, Vxworks etc.
  • JRTPLIB C++ object oriented RTP library
  • libRTP part of gnome-o-phone
  • libzrtpcpp - ZRTP extension library for ccRTP stack
  • LIVE.COM Streaming Media includes C++ RTP stack
  • oRTP Written in C, running on linux, win32 and arm-linux.
  • PJMEDIA: Small footprint media stack with a tiny RTP/RTCP stack suitable for DSP or embedded deployment
  • RTPlib C library
  • sipXmediaLib RTP + audio bridges, audio splitters, echo suppression, tone from generation (e.g. DTMF), streaming support, RTCP, G711 codecs, etc. from SIPfoundry
  • Secure RTP - see: SRTP
  • UCL Common Multimedia Library includes cross platform RTP stack
  • Vovida RTP Stack
  • 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.
  • Howler Technologies - optimised G.729 codec for softswitch market.
  • 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.
  • Vovida.org STUN server: A STUN server
  • 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

 

IVR platforms

  • Asterisk: Open Source PBX with built-in IVR server
  • Bayonne: GNU project IVR server
  • CT Server Perl based Open Source client/server library supporting Voicetronix Telephony hardware.
  • FreeSWITCH
  • OpenVXI: Implementation of VoiceXML
  • SEMS: Free/Open Source SIP media server with IVR capabilities
  • sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant)
  • YATE Yet Another Telephony Engine
  • See Also: VoiceXML

 

Voicemail servers

  • Asterisk: Open Source PBX with built-in Voicemail Server
  • FreeSWITCH
  • Lintad: Linux Telephone Answering Device - A Voice and Faxmail Server
  • OpenPBX: Open Source PBX with built in voicemail
  • OpenUMS: Linux Voicemail and Unified Messaging Server
  • SEMS: Free/Open Source SIP media server with built-in Voicemail and Voicebox Server
  • sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant)
  • VOCP: A Voicemail Server for voice modems
  • YATE Yet Another Telephony Engine with H.323, SIP and IAX support.

 

Speech

Text-to-speech and speech-to-text (voice recognition)

  • Festival: Voice synthesis system (implemented with a trainable neural network)
  • OpenSALT: Implementation of SALT
  • OpenVXI: Implementation of VoiceXML
  • Sphinx: speaker-independent speech recognizer
  • UniMRCP: cross-platform MRCP client and server

 

Fax Servers

 

Development platforms, protocol stacks

  • H323plus: Open Source H.323 Protocol Stack following on from the original openH323
  • OpenBloX: OpenBloX Open Source Java Diameter framework with all IMS and SIP servers interfaces; maintained by Traffix Systems,
  • OpenMGCP: Open Source MGCP Protocol Stack Developed with C and POSIX APIs,
  • OpenSS7SS7 Protocol Stack
  • ooh323c: Open Source 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

 

Billing

 

Codecs

 

Middleware

  • Ernie: Open Source Python based applications platform for VoIP and presence based applications
  • Mobicents: The most popular Open Source Service Logic Execution Environment (JSLEE) and SIP Application Server for the Java platform.
  • TALK: Web based CTI Solution (AJAX client) which provides call control, presence and directorty features.

 

Suite Solutions

  • Zoontelecom: Zoon Suite is a Open Source solution for make VoIP services with billing and more. (Spanish)

 

CTI Dialer utilities

    • Asterisk phonebook A common shared phone book directory for Asterisk PBX
    • 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.

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