VoIP Software Phones are basically a software for making VoIP calls using computer. Software phones are usually less expensive and it offers a better options for Computer Telephony Integration (CTI). In this section we have listed Free VoIP Software Phones that you can use in your office or at home.
List of VoIP Software Phones that you can use for making VoIP calls:
Ekiga (formely known as GnomeMeeting) is an open source VoIP and video conferencing application for GNOME. Ekiga uses both the H.323 and SIP protocols. It supports many audio and video codecs, and is interoperable with other SIP compliant software and also with Microsoft NetMeeting. After several weeks of hard work, we are pleased to announce Ekiga 2.0.2. This release only includes bug fixes and updated translations. Thanks to all the people involved in this release!
Speak Freely is a 100% software-based VoIP phone originally written in 1991 by John Walker, founder of Autodesk. After April of 1996, he discontinued development on the program. Since then, several other VoIP "phones" have cropped up all over the world. However, most of these programs cost money. Most of them have poor sound quality, and don't support some of Speak Freely's basic features such as encryption, the answering machine, or selectable compression.
Out of fustration in the shortcomings of the other programs, and even Speak Freely, I decided to implement what I thought were needed features myself, starting in August of 1997. I contacted John Walker, and he agreed to let me officially take over the project. In the spirit of Open Source SoftwareTM, I am licensing this program under the GNU General Public License.
Gspeakfreely is a VoIP system with a flexible component system. It implements a set of audio processing components which can be connected to each other or mixed together. The most important components are net in/output, which implement VoIP functionality and the OSS-DSP in/output component. Additionally there is a ISDN in/output component that allows making actual phone connections, and a file input component that can also play Internet radio streams. Also included is a fading plug-in, that can for example fade incoming calls into your music. New components can be developed for specific purposes, and combined with existing ones. The net in/output components also have conference support. The net input component can mix incoming audio data from different hosts.
Linphone is a web phone: it let you phone to your friends anywhere in the whole world, freely, simply by using the internet. The cost of the phone call is the cost that you spend connected to the internet.
Here are the main features of linphone:
1. Works with the Gnome Desktop under linux, (maybe some others Unixes, but this has never been tested). Nevertheless you can use linphone under KDE, of course !
2. Since version 0.9.0, linphone can be compiled and used without gnome, in console mode, by using the program called
3. Works as simply as a cellular phone. Two buttons, and one more to chat.
4. Linphones includes a large variety of codecs (G711-ulaw, G711-alaw, LPC10-15, GSM, SPEEX and iLBC ). Thanks to the Speex codec it is able to provide high quality talks even with slow internet connections, like 28k modems.
5. Understands the SIP protocol. SIP is a standardised protocol from the IETF (http://www.ietf.org), that is the organisation that made most of the protocols used in the internet. This guaranties compatibility with most SIP - compatible web phones.
6. You just require a soundcard to use linphone.
7. Other technical functionnalities include DTMF (dial tones) support though RFC2833 and ENUM support (to use SIP numbers instead of SIP addresses).
8. Linphone is free software, released under the General Public Licence.
9. Linphone is documented: there is a complete user manual readable from the application that explains you all you need to know.
10.Linphone includes a sip test server called "sipomatic" that automatically answers to calls by playing a pre-recorded message.
Minisip is a SIP User Agent ("Internet telephone").
It can be used to make phone calls, instant message and videocalls to your buddies connected to the same SIP network.
1. SIP compliant (RFC 3261 and more)
2. Multiple lines (users) on the same phone
3. Multiple incoming/outgoing calls simultaneously
4. Runs on multiple Operating Systems (Linux PC, Linux familiar IPAQ PDA, Windows XP and soon Windows Mobile 2003 SE)
5. Focus on security: TLS, end-to-end security, SRTP, MIKEY (DH, PSK, PKE)
6. Instant Messaging
7. Video conferencing
8. Spatial audio
9. Push-to-Talk (P2T)
10.Full Mesh audio conferencing
The OpenH323 project aims to create a full featured, interoperable, Open Source implementation of the ITU-T H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.
OpenH323 development is coordinated by Quicknet Technologies Inc. but is open to any interested party. Commercial and private use of the OpenH323 code, including use in commercial products and resale, is encouraged through use of the MPL (Mozilla Public license).
Open Source Contracting and Services
We are available to provide contract expertise in development, analysis or enhancement of your H.323 and VOIP products or projects. Recent past successes include projects for companies such as Agilent, Nortel, Lucent, Net2Phone and many others. By working with us you gain access to world class expertise, proven code bases utilized daily in 24x7 carrier operations. Further if your project require enhancements for actual services hosting our relationships can help you find the right relationship faster and more smoothly. Our service partners can provide:
1. Voice over IP Termination
2. Fax over IP delivery
3. MCU hosting and conferencing (including transcoding)
4. Video Conferencing
5. Automated Bulk Call Delivery
NetMeeting offers a complete Internet conferencing solution to improve business processes, help organizations to save time and money, and increase the productivity of their users.
More and more companies are adopting the use of Microsoft NetMeeting for its feature-rich conferencing capabilities. Businesses are choosing NetMeeting over competing products because it offers the most comprehensive, manageable solution for real-time communications and collaboration for use on their corporate intranet.
SIPSet is a SIP User Agent with a GUI front end that works with the Vovida SIP stack. You can use the SIPSet as a soft phone, to make and receives phone calls from your Linux PC.
This release implements these new features and functionality:
SIPSet can make calls through a SIP proxy.
SIPSet can register to receive calls through a SIP proxy.
SIPSet can make and receive calls directly with another User Agent
Quicknet Technologies is proud to announce the full release of our latest Internet SwitchBoard software - Version 4.0!
If you haven't experienced the latest upgrade during our recent Beta test you are missing out! Key features of this release include: access to multiple VoIP service providers using a single account; easy to use web-based account center; new calling capabilities including fax over IP using your standard fax machine (PC-to-a fax machine or multiple fax machines and fax to email addresses) and expanded international support with international ring tones, and localized dialing.
The Internet SwitchBoard 4.0 is a high performance Internet telephone application providing voice and fax calling capabilities taking full advantage of the advanced features of Quicknet's award winning hardware line. New features of the Internet SwitchBoard include:
** LEAST COST ROUTING - Calls are routed to the least cost Internet Telephone carrier OR your preferred carrier. Current carriers include BOTH Net2Phone and Deltathree.com who together serve over half the world's PC-to-Phone market.
** IP FAX CAPABILITY - Now you can plug in your fax machine and send faxes directly over the Internet! Send to another fax machine, to an email address or to a list of addresses using our Fax Broadcast feature!
** INCREASED RELIABILITY - Multiple service providers are accessed through a single account, dramatically improves overall reliability when you are making calls or sending faxes.
** SIMPLIFIED INTERFACE- The user interface, installation, registration and operation has been simplified for ease of use with out sacrificing capabilities.
** AUTO CONNECT - When a carrier becomes unavailable due to the Internet, our exclusive "Auto Connect" feature will re-route the call to the next available carrier guaranteeing a higher call completion rate for your calls.
Rs. 20,000 US$ 300
Today: Rs. 10,000 US$150
Course Duration: 30 hrs
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