login | register
Wed 01 of Aug, 2007 [09:10 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Shoutbox
  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.91s
  • Memory usage: 2.37MB
  • Database queries: 35
  • GZIP: Disabled
  • Server load: 3.14

Open Source VOIP Software

Open Source VOIP applications, both clients and servers.

Open source means all source code is available!! Do not post any "free but not open" software here!

SIP Proxies

  • Net-SIP A Perl SIP framework that includes a stateless proxy
  • 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 Proxy Server: Standards compliant sip proxy server
  • My SIP Switch: SIP Proxy server which allows using multiple SIP accounts with a single SIP login

SIP Clients (UA's)

Linux clients:

  • Cockatoo
  • Ekiga: SIP, H.323 audio and video softphone for various unices
  • 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
  • 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:

  • FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
  • PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
  • SFLphone, open-source multiplatform multi-protocol VoIP client
  • Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux

Windows clients:

  • 1videoConference alpha: a web2.0 VoIP video conferencing software for Asterisk.
  • 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
  • Linphone audio and video SIP softphone for Linux and Windows XP
  • MjUA: simple cross-platform SIP softphone, written in java, based on MjSip stack
  • Eyeball Messenger: Standards based soft client that is SIP and XMPP compliant
  • 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
  • wxCommunicator Windows softphone based on sipXtapi and wxWidgets 2.8.x, multi-account, conferencing, NAT support
  • YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.


SIP tools


SIP Protocol Stacks and Libraries

  • 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 written in C, and has language binding for Python.
  • 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.
  • Verona - GPL licenesed VOIP engine based on oSIP,eXosip,oRTP,ffmepg, works on Linux,Windows Mac-OS/X
  • PhClickDial - Verona based Active/X plugin for IE allowing ClickToDial functionallity

H.323 Clients

Linux clients:


MacOS X clients:


Windows clients:


H.323 Gatekeeper


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


RTP Protocol Stacks

  • JRTPLIB CUCL Common Multimedia Library includes cross platform RTP stack
  • oRTP Written in C, running on linux, win32 and arm-linux.
  • 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 suppression, 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.
  • MessengerSDK: VoIP, Video Telephony & IM SDK for PC, PDA and embedded platforme windowsXXXX, mobile, CE, Linux and uclinux
  • Voipong - Voice over IP (VoIP) sniffer and call detector.
  • NAT Traversal SDK: STUN, TURN and ICE development kit for integrating NAT Traversal into your application or device



PBX platforms

Some of these include SIP proxy functionality

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)
  • YATE Yet Another Telephony Engine
  • FreeSWITCH
  • See Also: VoiceXML



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)

Fax Servers


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 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.
  • OpenBloX: OpenBloX Open Source Java Diameter framework with all IMS and SIP servers interfaces; maintained by Traffix Systems,
  • OpenSS7: SS7 Protocol Stack


Radius Servers



Billing


Codecs


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



Created by oej, Last modification by Jaroslav Libak on Mon 30 of Jul, 2007 [23:14 UTC]

Comments Filter

multiple calls softphone

by Shengbin on Friday 16 of February, 2007 [12:21:35 UTC]
Hi,
I am doing research on VoIP now. I need a softphone to simulate multiple calls to evaluate the capacity of the VoIP network.
Can anybody suggest me a opensource softphone which can simulate multiple calls? It would be highly appreciated. My email address i s viruschidai@gmail.com.

multiple calls softphone

by Shengbin on Friday 16 of February, 2007 [12:20:54 UTC]
Hi,
I am doing research on VoIP now. I need a softphone to simulate multiple calls to evaluate the capacity of the VoIP network.
Can anybody suggest me a opensource softphone which can simulate multiple calls? It would be highly appreciated. My email address i s viruschidai@gmail.com.

multiple calls softphone

by Shengbin on Friday 16 of February, 2007 [12:20:41 UTC]
Hi,
I am doing research on VoIP now. I need a softphone to simulate multiple calls to evaluate the capacity of the VoIP network.
Can anybody suggest me a opensource softphone which can simulate multiple calls? It would be highly appreciated. My email address i s viruschidai@gmail.com.

multiple calls softphone

by Shengbin on Friday 16 of February, 2007 [12:20:24 UTC]
Hi,
I am doing research on VoIP now. I need a softphone to simulate multiple calls to evaluate the capacity of the VoIP network.
Can anybody suggest me a opensource softphone which can simulate multiple calls? It would be highly appreciated. My email address i s viruschidai@gmail.com.

VOIP BILLING SOLUTIONS Version VM3.9

by esha jones on Saturday 02 of December, 2006 [14:48:47 UTC]

re: VOIP BILLING SOLUTIONS Latest version VM 3.9


Hi,

We are offering different kinds of VOIp Solutions. The most latest version we are offering now is the VM 3.9.
This Version is 100% original copy.

We also offer 24x7 technical supports.

We also have complete package which include: VM software, SM, dialer, and webdesign.

For further information you need please email at: solutionsguru@gmail.com or chat live msn: solutionsguru@gmail.com


Thanks

Ms. Esha Jones


Fizan Telecom Iniviting Resellers from Gulf

by fizan telelcom on Friday 07 of July, 2006 [00:59:51 UTC]
We are pleased to introduce our Fizan Telecom Ltd . As Growing A2z Voip Provider having Our Server in cananda and Russia , we have high quality routes and high Perfoming Retail Plateform , Like " Calling cards , Sip Proxy Clients, Webcallback , Sms call back , Pc2phone , Whole sale termination , We are specialize in the Hot Routes from Gulf Area, Like India , Pakistan , Nepal , Egypt , Bangladesh , Philpine , Indonesia , ,Sri-Lanka ,
We warmly Inivites resellers from All over the world .
Regards
Mr.Ali
Marketing Dept www.fizantel.com
info@fizantel.com
fizatel@hotmail.com
fizatel@yahoo.com

Fizan Telecom A2z Provider Inviting Resellers

by fizan telelcom on Friday 07 of July, 2006 [00:59:22 UTC]
We are pleased to introduce our Fizan Telecom Ltd . As Growing A2z Voip Provider having Our Server in cananda and Russia , we have high quality routes and high Perfoming Retail Plateform , Like " Calling cards , Sip Proxy Clients, Webcallback , Sms call back , Pc2phone , Whole sale termination , We are specialize in the Hot Routes from Gulf Area, Like India , Pakistan , Nepal , Egypt , Bangladesh , Philpine , Indonesia , ,Sri-Lanka ,
We warmly Inivites resellers from All over the world .
Regards
Mr.Ali
Marketing Dept www.fizantel.com
info@fizantel.com
fizatel@hotmail.com
fizatel@yahoo.com

Fizantel Offer Special Prices In hote Route

by fizan telelcom on Saturday 01 of July, 2006 [00:49:06 UTC]
We are pleased to introduce our Fizan Telecom Ltd . A growing Voip

Provider having Our Server in cananda and Russia ,
we have high quality routes and high Perfoming Retail Plateform , Like

" Calling cards , Sip Proxy Clients, Webcallback , Sms call back ,

Pc2phone , Whole sale termination ,
We warmly Inivites resellers from All over the world .Fizantel Offer Special Prices In hote Routes , Escpecially , India , pak , Nepal , Sri-lanka , Egypt , bangladesh , Phipine , Saudia ,Indonesia
Regards
info@fizantel.com
fizatel@hotmail.com
fizatel@yahoo.com

www.fizantel.com

by fizan telelcom on Saturday 01 of July, 2006 [00:47:26 UTC]
We are pleased to introduce our Fizan Telecom Ltd . A growing Voip

Provider having Our Server in cananda and Russia ,
we have high quality routes and high Perfoming Retail Plateform , Like

" Calling cards , Sip Proxy Clients, Webcallback , Sms call back ,

Pc2phone , Whole sale termination ,
We warmly Inivites resellers from All over the world .
Regards
www.fizantel.com
info@fizantel.com
fizatel@hotmail.com
fizatel@yahoo.com

Fizan Telecom Iniviting Resellers

by fizan telelcom on Saturday 01 of July, 2006 [00:46:19 UTC]
We are pleased to introduce our Fizan Telecom Ltd . A growing Voip

Provider having Our Server in cananda and Russia ,
we have high quality routes and high Perfoming Retail Plateform , Like

" Calling cards , Sip Proxy Clients, Webcallback , Sms call back ,

Pc2phone , Whole sale termination ,
We warmly Inivites resellers from All over the world .
Regards
info@fizantel.com
fizatel@hotmail.com
fizatel@yahoo.com

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:

Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver