VOIP开源软件列表

通用型

• GNU/Unesco Software Directory : Telephony
• Open Source SIP and Media Links
• SIPfoundry: Organzation for development of Open Source VOIP Software, founded by Pingtel in cooperation with Vovida.org and the reSIProcate community

SIP Proxies 代理

• sipd SIP Proxy
• SIP Express Router (SER): the SIP router/proxy/jack-in-all-trades from IPtel.org
• partysip
• SaRP SIP and RTP Proxy in Perl
• Siproxd SIP and RTP Proxy
• 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
• JAIN-SIP Proxy
• Mini-SIP-Proxy A very tiny perl POE based SIP proxy
• OpenSER: GPL SIP Server with TLS support
• MjServer: cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack
• OpenSBC: MPL licensed SIP proxy/registrar/B2BUA with NAT traversal and ENUM

SIP Clients (UA's) 客户端

Linux clients:

• Cockatoo
• Ekiga: SIP, H.323 audio and video softphone for various unices
• Kphone
• Linphone
• 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
• PhoneGaim
• PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
• SFLphone, open-source multiplatform multi-protocol VoIP client
• OpenWengo: a fully SIP compliant multiplatform softphone with many features
• OpenZoep: GPL telephone and IM messaging client engine
• Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
• sipXphone from SIPfoundry, previously known as the Pingtel phone
• sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
• Twinkle
• YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
• YeaPhone: A SIP softphone for the Yealink USB-P1K handset based on the libLinphone backend
• FreeSWITCH
• http://www.opensipstack.org MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.

 

MacOS X clients:

• PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
• FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk

 

Windows clients:

• FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
• 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
• OpenWengo: a fully SIP compliant multiplatform softphone with many features
• OpenZoep: GPL telephone and IM messaging client engine
• PhoneGaim
• PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
• SIP COMMUNICATOR Java based softphone
• Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
• sipXphone from SIPfoundry, previously known as the Pingtel phone
• http://www.opensipstack.org MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
• sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
• YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.

SIP tools 工具

• Callflow: Generates SIP Call Flow diagrams
• SIP-CallerID: SIP Caller ID retrieval and lookup
• SIPbomber: SIP proxy testing tool
• Sipp: SIP performance tester
• pjsip-perf: SIP transaction and call performance measurement tool
• Sipsak: SIP testing tool
• Vovida.org load balancer: SIP Load Balancer
• 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 Protocol Stacks and Libraries 协议栈和库

• 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.
• MjSip - complete and powerful java-based SIP library for both J2SE and J2ME platforms.
• oSIP Library SIP Library
• eXosip - eXtended osip library
• Vovida SIP Vovida SIP stack
• reSIProcate SIP stack and sample Application from SIPfoundry
• NIST SIP Various SIP appications and tools in Java
• PJSIP: Small footprint, high performance, and ultra-portable SIP stack in C.
• Twisted Python protocol stacks and applications includes SIP support
• OSP client protocol stack and SIPfoundry
• libdissipate SIP stack
• sipXtackLib an RFC 3261, 3263 complient SIP stack from SIPfoundry
• minisip includes a SIP stack
• http://sofia-sip.sourceforge.net Sofia-Sip is SIP stack implementation with STUN and presense support
• http://www.opensipstack.org MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.

H.323 Clients

Linux clients:

• Ekiga
• GnomeMeeting
• YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
• FreeSWITCH

MacOS X clients:

• ohphoneX
• FreeSWITCH

Windows clients:

• OpenPhone
• FreeSWITCH

Not OpenSource - Pending Removal Neos

H.323 Gatekeeper

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

IAX clients

• IAXComm for Linux, MacOS X and Windows
• Kiax - for Linux (QT3) and Windows (QT4), based on iaxclient, GPL
• QtIax from http://www.holgerschurig.de/qtiax.html
• SFLphone, open-source multiplatform multi-protocol VoIP client (IAX support is planned)
• MozIAX
• YakaPhoneSimple, Free, Open Source, Skinnable IAX/IAX2 Softphone from YakaSoftware
• YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
• FreeSWITCH

RTP Proxies

• Maxim Sobolev's RTPproxy: Works with SIP express router to traverse NAT, also functions as RTP gateway between IPv4 and IPv6
• AG Projects: SER MediaProxy works with SIP express router, has load-balancing using DNS SRV records and accounting capabilities

RTP Protocol Stacks

• JRTPLIB cUCL Common Multimedia Library inlcudes cross platform RTP stack 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.
• oRTP Written in C, based on glib for unix and windows portability
• ccRTP C++ library based on GNU Common C++
• LIVE.COM Streaming Media includes C++ RTP stack
• Vovida RTP Stack
• RTPlib C library
• libRTP part of gnome-o-phone
• sipXmediaLib RTP + audio bridges, audio splitters, echo supression, tone from generation (e.g. DTMF), streaming support, RTCP, G711 codecs, etc. from SIPfoundry
• Secure RTP - see;"> SRTP
• YRTP - Yate RTP stack, that can be used in other projects.
• FreeSWITCH
• PJMEDIA: Small footprint media stack with a tiny RTP/RTCP stack suitable for DSP or embedded deployment

Other tools

• Vovida.org STUN server: A STUN server
• Vomit converts a Cisco IP phone conversation (recorded with TCPdump) into a standard WAV file
• Oreka capture and retrieval of SIP, Cisco Skinny (SCCP) and raw RTP sessions with audio compression, rdbms metadata storage and web based user interface.
• MORCC - automated online Calling Card store. Paypal integrated.

PBX platforms

Some of these include SIP proxy functionality

• Asterisk: Open Source PBX. Supports IAX, SIP, MGCP, H.323 and other protocols
• Asterisk Business Edition
• OpenPBX: Open Source PBX developed using Perl
• PBX4Linux: ISDN PBX with H.323 GW
• sipX - The SIP PBX for Linux from SIPfoundry, sipX on freshmeat.net
• SIPexchange PBX Pingtel's SIP PBX
• YATE Yet Another Telephony Engine - supports H.323, SIP, IAX, PSTN
• FreeSWITCH

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.
• OpenVXI: Implementation of VoiceXML
• sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant)
• See Also: VoiceXML
• YATE Yet Another Telephony Engine
• FreeSWITCH

Voicemail servers

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

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
• FreeSWITCH

Fax Servers

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

Development platforms, protocol stacks

• 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

Radius Servers

• BSDRadius: Radius server for VoIP

Billing

• See Open Source Billing Systems

Codecs

• See Codec Software

Middleware

• Mobicents: The most popular Open Source Service Logic Execution Environment (JSLEE) and SIP Application Server for the Java platform.
• Ernie: Open Source Python based applications platform for VoIP and presence based applications

 

http://download.youkuaiyun.com/source/627469
http://wellya.net/read-htm-tid-9548.html
http://www.ctiforum.com/voip.html
http://hi.baidu.com/rbtk/blog/item/c9d6d0c89f3be6177f3e6fbd.html