Tuesday, November 30, 2010

Ekiga is an H.323 compatible videoconferencing and VOIP/IP-Telephony application that allows you to make audio and video calls to remote users with H.323 hardware or software (such as Microsoft Netmeeting). It supports all modern videoconferencing features, such as registering to an ILS directory, gatekeeper support, making multi-user conference calls using an external MCU, using modern Quicknet telephony cards, and making PC-To-Phone calls.

Ekiga was previously known as GnomeMeeting.

Twinkle is a soft phone for VoIP communcations using the SIP protocol. You can use Twinkle for direct IP phone to IP phone communications or in a network using a SIP proxy to route your calls.

In addition to making basic voice calls, Twinkle also provides the following features:

  • 2 call appearances (lines)
  • Multiple active call identities
  • Custom ring tones
  • Call Waiting
  • Call Hold
  • 3-way conference calling
  • Mute
  • Call redirection on demand
  • Call redirection unconditional
  • Call redirection when busy
  • Call redirection no answer
  • Reject call redirection request
  • Blind call transfer
  • Reject call transfer request
  • Call reject
  • Repeat last call
  • Do not disturb
  • Auto answer
  • User definable scripts triggered on call events
  • E.g. to implement selective call reject or distinctive ringing
  • RFC 2833 DTMF events
  • Inband DTMF
  • Out-of-band DTMF (SIP INFO)
  • STUN support for NAT traversal
  • Send NAT keep alive packets when using STUN
  • NAT traversal through static provisioning
  • Missed call indication
  • History of call detail records for incoming, outgoing, successful and missed calls
  • DNS SRV support
  • Automatic failover to an alternate server if a server is unavailable
  • Other programs can originate a SIP call via Twinkle, e.g. call from address book
  • System tray icon
  • System tray menu to quickly originate and answer calls while Twinkle stays hidden
  • User defineable number conversion rules

WengoPhone is a SIP phone which allows users to speak at no cost from one's computer to other users of SIP compliant VoIP software. It also allows users to call landlines, cellphones, send SMS messages and to make video calls. None of this functionality is tied to a particular SIP provider and can be used with any provider available on the market, unlike proprietary solutions such as Skype.

Speak Freely is a 100% free Internet telephone originally written in 1991 by John Walker, founder of Autodesk. After April of 1996, he discontinued development on the program. Since then, several other Internet "telephones" 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 Speak Freely's basic features such as encryption, the answering machine, or selectable compression.

Gspeakfreely is a VoIP system with a flexible component system. It implements a set of audio processing component 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 SIP webphone with support for several different codecs, including speex.

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.

linphone features include:

  • Works with the Gnome Desktop under Linux, (maybe others Unixes as well, but this has never been tested). Nevertheless you can use linphone under KDE, of course!
  • Since version 0.9.0, linphone can be compiled and used without gnome, in console mode, by using the program called "linphonec"
  • Works as simply as a cellular phone. Two buttons, no more.
  • Linphones includes a large variety of codecs (G711-ulaw, G711-alaw, LPC10-15, GSM, and SPEEX). Thanks to the Speex codec it is able to provide high quality talks even with slow internet connections, like 28k modems.
  • Understands the SIP protocol. SIP is a standardised protocol from the IETF, that is the organisation that made most of the protocols used in the Internet. This guaranties compatibility with most SIP - compatible web phones.
  • You just require a soundcard to use linphone.
  • Other technical functionalities include DTMF (dial tones) support though RFC2833 and ENUM support (to use SIP numbers instead of SIP addresses).
  • Linphone is free software, released under the General Public Licence.
  • Linphone is documented: there is a complete user manual readable from the application that explains you all you need to know.
  • Linphone includes a sip test server called "sipomatic" that automatically answers to calls by playing a pre-recorded message.

minisip is a SIP VoIP soft phone that implements additional security features such as mutual authentication, encryption and integrity of on-going calls, and encryption of the signaling (SIP over TLS). These security features use work-in-progress IETF standards (SRTP and MIKEY).

OhPhone is a H.323 Video Conferencing Program compatible with other H.323 video conferencing programs including Microsoft NetMeeting.

OhPhone supports full duplex audio and bi-directional video. It requires a full duplex sound card for audio support and a Bt848/878 based video card (using the bktr driver) for video capture.

OhPhone uses the OpenH323 and PWLib libraries, developed by Equivalence Pty.

NetMeeting is Microsoft's free H.323-compliant VoIP software phone for Windows.

The Internet SwitchBoard software is the client software for MicroTelco services and is included with the purchase of the Internet PhoneJACK or Internet PhoneCARD.

The Internet Switchboard was designed to be used with Quicknet hardware and a MicroTelco Services account. The Internet SwitchBoard can be configured with your firewall and features voice control with worldwide phone and dial tone emulation.

The Internet SwitchBoard software is a PC-to-PC, PC-to-Phone, Fax-to-Email, and Fax-to-Fax calling application that allows users to make low cost calls worldwide to other phones or fax machines.

PC-to-Phone and Fax-to-Fax calls are as easy to dial as using a phone or fax machine. PC-to-PC calls are made by dialing an IP address and are free. FAX-to-Email documents are electronically transmitted as virus free e-mail attachments and are free if sent individually. Recipients can view files in popular e-mail clients.

Internet Switchboard features include:

  • Low calling rates through MicroTelco Services
  • Auto call connect - automatic connection and least cost routing feature that connects your call using the next available carrier when the chosen carrier is unavailable
  • Least cost routing - for voice amongst leading global IP carriers
  • Automatic firewall detection
  • Automatic fax detection - allowing a fax machine to be plugged into a compatible card using the Internet SwitchBoard and route faxes to email or another fax machine via the Internet
  • International phone emulation's & connectivity
  • Low account balance warning
  • Call connect announcement
  • Auto gain control
  • Supports any type of Internet connection, including broadband
  • Microsoft Operating support including Windows 98/98SE, ME, 2000, and Windows XP

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.

The current release of SIPSet implements these 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.

KPhone is a SIP User Agent for Linux. It implements the functionality of a VoIP Softphone but is not restricted to this. KPhone is licensed under the GNU General Public License. KPhone is written in C++ and uses Qt.

Jabbin is an open source Jabber client program that allows free PC to PC calls using VoIP over the Jabber network.


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