woojar ? VOIP开源列表

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VOIP开源列表

Filed under: Tech articles — Tags: open source, voip — woojar @ 2:53 pm

[tag]voip, open source[/tag]

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)

    1 Comment »

    1. Hi, good post. I have been wondering about this issue,so thanks for posting.

      Comment by AndrewBoldman — June 4, 2009 @ 9:23 pm

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