Sip Softphone and Ekiga Talk Multimedia Communicator Software for Windows

Voip Ekiga Communicator
$0 - 3DMail

callbackbuttun is also a very clevere alternative to classical 0800 it the service number. Because a 0800'er call number costs not only uniquely (only alone the dispatching costs 62.50 euro with the federal net agency), but the telefongesellschaften takes also not insignificant monthly basic charges and to it still quite expensive Verbindnugsentgelte around the discussions to you to transport. The Rueckrufbutton is available however without basic charges (tariff Classic) and without Setupgebuehr. Only the connecting costs result, which are however favorable and are by far under the costs of the supply of a 0800er. The Rueckrufbutton is however not only an alternative to 0800er numbers, but offers further strong advantages for your homepage. Announce, insert and loose-put. Starting assets give it free of charge in addition so that you can test and telephone immediately. Further features: * Simple and fast configuration * Restriction on portable radio or international call numbers * Adjustable opening times * Cost control by SiPHome Prepaid and single connecting proof * Starting assets for immediate trying out! * Adjustment to own layout by Stylesheet file * Integration of personal announcement and own diagram * In several languages (German and English) * own telephone number, SIP - IAX address as goal call number definably * For programmer even only the API usable, in order to be able to start e.g. calls from your own software

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Phoner
$0 - Heiko Sommerfeldt

Phoner is a free softphone for ISDN and VoIP telephony. An answering machine and Text-To-Speech support complete the large package.

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Axon Virtual PBx System
$0 - NCH Swift Sound Software

Axon is a virtual PBX designed to manage calls within a business or call center environment. Any business, whether small or large, can now implement a scalable PBX solution with an existing PC which connects to phone lines and extensions using VoIP technology.
Offering all the normal features of a traditional PBX such as allowing internal or external calls and more advanced call queuing for call center applications the software routes all calls within a business.

Features
~ Manages telephone calls within a business.
~ All routing of calls is digital over the computer network (or even a wireless network).
~ Unlimited number of extensions or external lines.
~ Supports all the standard PBX features like call transfer, hold, and record.
~ Includes a call queue sequencer with voice prompting and on-hold messages player (links to the IMS On-Hold Messages Player).
~ Uses the open SIP standard for signalling (RFC 3261) so it works with the broad range of SIP compatible phones and other devices.
~ Works with hardware type VoIP extensions which just plug into the LAN (eg. Grandstream, Sipura etc.).
~ Works with computer based softphones (eg. the Express Talk Softphone) where you make calls through your computer.
~ Works with USB phones plugged into your computer.
~ Links directly to VoIP services for your external lines (with cheaper calls). See Recommended VoIP Gateways for more information.
~ If no VoIP services are available can also link to ordinary external lines with FXO adapters.
~ Automatic "one-click" installer to setup the PBX in just minutes. Does not require detailed technical setup.
~ All configuration such as adding extensions is done using a web-like control panel through your internet browser so it can be done remotely.

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VoIP SIP SDK for .NET and Win32 COM
$999 - conaito Technologies

VoIP SIP SDK - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

Our brand-new SIP SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications.
It accelerates the development of SIP/ RTP compliant soft phone with
a fully-customizable user interface and brand name. The conaito VoIP SIP SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either by speaking and/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller, acoustic echo cancellation, noise cancellation, reverb cancellation and Voice activity detection.

Here is a list of the main features of the conaito VoIP SIP client:
* Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law and iLBC Codec).
* Registration on SIP Server (SIP Registrar).
* Instant text messaging.
* Microphone and Speaker Visualization support.
* Microphone and Speaker Volume with Mute support.
* Audio device selection.
* Fully-customizable user interface.
* Packetloss resistant (by using iLBC codec).
* Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
* Works with all kind of Internet connections.
* Royalty free licensing
* No Yearly/Monthly fee
* Very easy to incorporate
* VAD (Voice activity detection), Reverb, Echo and Noise cancellation or suppression, AGC (auto gain controller).

And much more! Try it today.

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SIP SDK
$999 - conaito Technologies

SIP SDK - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

Our brand-new SIP SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications.
It accelerates the development of SIP/ RTP compliant soft phone with
a fully-customizable user interface and brand name. The SIP SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either by speaking and/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller, acoustic echo cancellation, noise cancellation, reverb cancellation and Voice activity detection.

Here is a list of the main features of the conaito VoIP SIP client:
* Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law and iLBC Codec).
* Registration on SIP Server (SIP Registrar).
* Instant text messaging.
* Microphone and Speaker Visualization support.
* Microphone and Speaker Volume with Mute support.
* Audio device selection.
* Fully-customizable user interface.
* Packetloss resistant (by using iLBC codec).
* Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
* Works with all kind of Internet connections.
* Royalty free licensing
* No Yearly/Monthly fee
* Very easy to incorporate
* VAD (Voice activity detection), Reverb, Echo and Noise cancellation or suppression, AGC (auto gain controller).

And much more! Try it today.

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SIP DLL
$999 - conaito Technologies

SIP DLL - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

Our brand-new SIP DLL provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications.
It accelerates the development of SIP/ RTP compliant soft phone with
a fully-customizable user interface and brand name. The SIP DLL contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either by speaking and/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller, acoustic echo cancellation, noise cancellation, reverb cancellation and Voice activity detection.

Here is a list of the main features of the conaito SIP DLL:
* Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law and iLBC Codec).
* Registration on SIP Server (SIP Registrar).
* Instant text messaging.
* Microphone and Speaker Visualization support.
* Microphone and Speaker Volume with Mute support.
* Audio device selection.
* Fully-customizable user interface.
* Packetloss resistant (by using iLBC codec).
* Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
* Works with all kind of Internet connections.
* Royalty free licensing
* No Yearly/Monthly fee
* Very easy to incorporate
* VAD (Voice activity detection), Reverb, Echo and Noise cancellation or suppression, AGC (auto gain controller).

And much more! Try it today.

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