开源sip server & sip client 和开发库 一览

标签：func   into   ipa   remote   ips   license   interface   page   osi   

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 || SIP, H.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
• Shtoom: SIP 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
• Shtoom: SIP 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 || SIP, H.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
• Shtoom: SIP 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:

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

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 || SIP, H.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

• GNU Gatekeeper - for Linux, Windows, Mac etc.

IAX clients

• FreeSWITCH
• IAXComm for Linux, MacOS X and Windows
• Kiax - for Linux, Windows and MacOS, based on iaxclient, GPL
• MozIAX
• QtIax from http://www.holgerschurig.de/qtiax.html
• SFLphone, open-source multiplatform multi-protocol VoIP client (IAX support is planned)
• YakaPhoneSimple, Free, Open Source, Skinnable IAX/IAX2 Softphone from YakaSoftware
• YateClient is multiprotocol and multiplatform softphone with H.323, SIP and IAX support.

RTP Proxies

• AG Projects: MediaProxy 1 works with SIP express router and OpenSER, has load-balancing using DNS SRV records and accounting capabilities
• Maxim Sobolev RTPproxy: Works with SIP express router to traverse NAT, also functions as RTPgateway between IPv4 and IPv6
• MediaProxy 2 is more scalable using kernel space switching and works with OpenSIPs

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

• MSRPRelay from AG Projects

XCAP servers

• OpenXCAP from AG Projects

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

• Asterisk: Open Source PBX. Supports IAX, SIP, MGCP, H.323 and other protocols
• CallWeaver: a fork of Asterisk with T.38 termination
• FreeSWITCH Open Source PBX and Soft Switch
• OpenPBX: Open Source PBX developed using Perl
• PBX4Linux: ISDN PBX with H.323 GW
• SIPexchange PBX Pingtel‘s SIP PBX
• sipwitch: GNU project‘s Pure SIP call server, sipwitch on freshmeat.net
• sipX - The SIP PBX for Linux from SIPfoundry, sipX on freshmeat.net
• SIP - It‘s the Rage! - Rage! Business Office Xchange based on SipFoundry
• YATE Yet Another Telephony Engine - supports H.323, SIP, IAX, PSTN

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

• Asterisk Fax Email Gateway
• Lintad: Linux Telephone Answering Device - A Voice and Faxmail Server
• Hylafax

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,
• OpenSS7: SS7 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

• Aradial: Radius server and Billing for VoIP
• BSDRadius: Radius server for VoIP
• Interlink RADIUS Server RADIUS Server Software
• RadBox RADIUS Server + Billing System. (For a work, you nead instal Framework 2.0)

Billing

• See Open Source Billing Systems

Codecs

• See Codec Software

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.

开源sip server & sip client 和开发库 一览

标签：func   into   ipa   remote   ips   license   interface   page   osi   

原文地址：http://www.cnblogs.com/lidabo/p/7587709.html