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Fri 09 of May, 2008 [16:55 UTC]

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  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
  • Christopher Faust, Wed 07 of May, 2008 [15:28 UTC]: When I try to startx I ge input not supported. Though before installing asterisk I had no video issue to start the GUI
  • Christopher Faust, Wed 07 of May, 2008 [15:26 UTC]: Hi Nick, I got centos 5.1 and asterisk up But now I cannot start startx I have set the depth from 24 to 16 for the video i810 driver for the i845 on my netvista machine but I cannot start GNOME. Please advise
  • Nick Barnes, Wed 07 of May, 2008 [10:01 UTC]: Howard - You'll need to provide a lot more information if you really want help.
  • Nick Barnes, Wed 07 of May, 2008 [10:00 UTC]: Christopher - Search the Wiki and you'll find a page I wrote detailing exactly what you have to do for Asterisk 1.4 + CentOS 5.1.
Server Stats
  • Execution time: 0.52s
  • Memory usage: 2.38MB
  • Database queries: 36
  • GZIP: Disabled
  • Server load: 0.81

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
  • MySIPSwitch: SIP Proxy server which allows using multiple SIP accounts with a single SIP login
  • SIPVicious tool suite: tools for auditing sip devices



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
  • SippySkype from mhspot.com Skype SIP UA - Multiplatform - Open Source
  • 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
  • SippySkype from mhspot.com Skype SIP UA - Multiplatform - Open Source

Windows clients:

  • VMukti (formerly 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
  • SippySkype from mhspot.com Skype SIP UA - Multiplatform - Open Source
  • 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

  • Aloha Spring based J2SE SIP A/S which leverages optimistic concurrent model and supports multiple persistence models
  • 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.
  • Voipong - Voice over IP (VoIP) sniffer and call detector.




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
  • H323plus: Open Source H.323 Protocol Stack following on from the original openH323
  • 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

Suite Solutions

  • Zoontelecom: Zoon Suite is a Open Source solution for make VoIP services with billing and more. (Spanish)


Created by oej, Last modification by Robbie Clutton on Sat 03 of May, 2008 [22:27 UTC]

Comments Filter

by linkx on Friday 22 of February, 2008 [07:33:42 UTC]






























































































































VoIP Security Solutions

by jenniferhan on Thursday 27 of December, 2007 [06:56:20 UTC]
SpeedVoIP is a professional VoIP Security and VoIP anti blocking solutions provider.
The core solution for VoIP Security and VoIP anti-blocking is VGCP (VoiceGuard Control Protocol).
It can work with any 3rd-party Softphone / ATA / Gateway / IP Phone / IADs and SIP proxy or server.
It can work in the way similar to that of SOHO router, but it only encrypts and decrypts SIP and RTP packets on link layer, not to handup these packets to IP stack for forwarding while bypassing other data packets originating from SIP terminals. In this scenario, peak throughput and minimal CPU overhead can be easily achieved.

VoiceGuard can real-time incorporate light-weight traffic for puzzling and bypassing VoIP blocking system without consuming more bandwidth and compromising voice quality. Even in some circumstance, VoiceGuard can simulate traffic behavior of universal data networking protocol such as OICQ, MSN and so on.

For more information, please refer to: http://www.speed-voip.com/index-36.html

Andy
xd.wong@speed-voip.com
andywong-01@hotmail.com

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.

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


voice quality

by Pawel on Tuesday 18 of April, 2006 [19:46:38 UTC]
Hi !
Does anyone know free software, to measure voice quality in MOS scale (P.800, PSQM, or whatever)? I spent a lot of time on google but didn't find anything free :(

we pay

by reza on Saturday 04 of February, 2006 [12:22:28 UTC]
we are looking for a linux based programmer for improving our SIP proxy Project contact me at grkashani@yahoo.com

Need SIP Softphone and we pay

by Yeaw Sing on Wednesday 11 of January, 2006 [11:51:13 UTC]
I need a SIP softphone for my Asterisk server. If anyone have develop a SIP softphone or would like to earn $$$ for developing SIP Softphone please contact me on yeawsing@gmail.com

setup Asterisk Prepadi Calling card

by Dirgantara on Wednesday 11 of January, 2006 [07:52:20 UTC]
hi i need help
and some body can teach me how to create asterisk with mysql as prepaid calling card
current iam already install asterisk with mysql cdr and Asterisk Management Portal.
other questions how to setup like SIP.conf in AMP ?, if iam using ExpressTalk Softphones any sample conf in mysq DB ?

thanks

Re: SIP-communicator.org

by marco on Friday 14 of October, 2005 [13:45:30 UTC]
i need modify the code of sip source (client).can i use SIP-communicator or i must use GPL sip source??

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