sip相关玩意
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Asterisk
SIP Express Router (SER):
VOIP开源软件列表
通用型
MacOS X clients:
Windows clients:
H.323 Gatekeeper
- GNU/Unesco Software Directory : Telephony
- Open Source SIP and Media Links
- SIPfoundry: Organzation for development of Open Source VOIP Software, founded byPingtel in cooperation with Vovida.org and the reSIProcate community
- 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 fromSIPfoundry
- 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
- 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") fromSIPfoundry based onsipXtapi
- 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") fromSIPfoundry based onsipXtapi
- YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
- 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 bySIPfoundry
- 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 fromSIPfoundry
- 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 andSIPfoundry
- libdissipate SIP stack
- sipXtackLib an RFC 3261, 3263 complient SIP stack fromSIPfoundry
- 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.
- Ekiga
- GnomeMeeting
- YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
- FreeSWITCH
- ohphoneX
- FreeSWITCH
- OpenPhone
- FreeSWITCH
H.323 Gatekeeper
- GNU Gatekeeper - for Linux, Windows, Mac etc.
- 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
- Maxim Sobolev's RTPproxy: Works withSIP express router to traverse NAT, also functions asRTP gateway between IPv4 and IPv6
- AG Projects: SER MediaProxy works withSIP express router, has load-balancing using DNS SRV records and accounting capabilities
- JRTPLIB cUCL Common Multimedia Library inlcudes cross platform RTP stack library forskype 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. fromSIPfoundry
- 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
- Vovida.org STUN server: ASTUN 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.
- Asterisk: Open Source PBX. SupportsIAX,SIP, MGCP, H.323 and other protocols
- Asterisk Business Edition
- OpenPBX: Open Source PBX developed usingPerl
- PBX4Linux: ISDN PBX with H.323 GW
- sipX - The SIP PBX for Linux fromSIPfoundry,sipX on freshmeat.net
- SIPexchange PBXPingtel's SIP PBX
- YATE Yet Another Telephony Engine - supportsH.323,SIP, IAX, PSTN
- FreeSWITCH
- 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 ofVoiceXML
- 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
- 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
- Festival: Voice synthesis system (implemented with a trainable neural network)
- OpenSALT: Implementation ofSALT
- OpenVXI: Implementation ofVoiceXML
- Sphinx: speaker-independent speech recognizer
- FreeSWITCH
- Lintad: Linux Telephone Answering Device - A Voice and Faxmail Server
- Hylafax
- Asterisk Fax Email Gateway
- 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
- BSDRadius: Radius server for VoIP
- See Open Source Billing Systems
- See Codec Software
- 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
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