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)
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Hi, good post. I have been wondering about this issue,so thanks for posting.
Comment by AndrewBoldman — June 4, 2009 @ 9:23 pm