Best softphone software for free download
MAX Phone 1.2
MAX Phone is the latest VoIP application from MAIN Telecom, allowing users to enjoy rich real-time communications.
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VaxVoIP SIP activeX SDK 1.0
VaxVoIP SIP SDK provides tools and components to quickly add SIP (Session Initiation Protocol) based IP-Telephony make and receive phone calls feature in your web pages and software applications. You can easily develope net-to-phone apps & web pa
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One4000 SoftGlobalPhone 1.0
One4000 SoftGlobalPhone, a Phone-in-the-Phone that lets you take advantage of free broadband calling and super low worldwide rates without changing your operator. 9 ways of calling from all of your phones, including PDA, cellulars and your PC.
Related software: mobile, telephony, voip, communications, sip, softphone, callback, …
Adore Softphone 2.0
Adore SIP Softphone comes with an intuitive Graphical User Interface and can be easily customized. It's designed with SIP technology that can be used to make outbound phone calls from PC to any telephone.
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VoIP SIP SDK with DLL, ActiveX and .NET 2.5.2
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 and websites. 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 suppression, noise suppression, reverb suppression and mute microphon Key Features * 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, Speex, GSM6.10, iLBC, L16, g723 and g729 Codec) * Encrypt SIP account settings (protect your SIP account settings in websites) * Secure Weblicensing (protect your license in websites) * Multi-User conference support * Multi-line (simultaneous calls) support (Multiple Concurrent calls) * Call Hold support * Call Transfer support * Instant text messaging (MIME) support and typing indication * Mute microphone/speaker for each line * DNS SRV resolution for SIP servers (RFC 3263) * Stereo codec (L16) * RTCP * Auto-answer * Do Not Disturb (DND) * Adaptive jitter buffer * Adaptive silence * Advanced configurable digital voice processing features ...and much more. Try it today!
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SIP Phone DLL 3.0
SIP Phone DLL for .NET and ActiveX - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications. Our brand-new SIP Phone 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 and websites. It accelerates the development of SIP/ RTP compliant soft phone with a fully-customizable user interface and brand name. The conaito SIP Phone 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 suppression, noise suppression, reverb suppression and mute microphon Key Features * 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, Speex, GSM6.10, iLBC, L16, g723 and g729 Codec) * Encrypt SIP account settings (protect your SIP account settings in websites) * Secure Weblicensing (protect your license in websites) * Multi-User conference support * Multi-line (simultaneous calls) support (Multiple Concurrent calls) * Call Hold support * Call Transfer support * Instant text messaging (MIME) support and typing indication * Mute microphone/speaker for each line * DNS SRV resolution for SIP servers (RFC 3263) * RTCP * Auto-answer * Do Not Disturb (DND) * Adaptive jitter buffer * Adaptive silence * PLC (Packet Lost Concealment) * AGC, VAD, AEC, Noise Concealment ...and much more. Try it today!
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MAXIVR Studio 3003
MAX IVR Studio Features: Performance and Effectiveness: * Call key select menus and ability for caller to enter numbers (data entry). * Unlimited (selectable) out-going messages and menus. * Fully programmable setup with examples. * call transfer (subject to your company PBX features). * call simulator is included so you can test your system off-line.</li> * extended message play on key press including support for wav , GSM and a number of other ormats. * Text-to-Speech voice synthesis as an alternative to recording or importing wav or GSM files. * ability to access any database through ODBC.</li> * support socket programming where can be sent/receive data to/from another PCs.</li> * support broadcasting actions such as send Email, Fax and SMS.</li> * ability to record message or record all the calls. * support any arithmetic operations. * ability to edit the windows registry. * ability to import external files or run other software to process data or report information. Reliability, Scalability, and Flexibility * easy to use, graphical representation of common tape-deck controls, Pull-down menus and extensive on-line help files that greatly * reduce the need for costly user training * visual displays of audio waveforms make it easy to see the effects of editing * open development and editing environment provides choice and flexibility in importing and exporting audio files in a variety of formats import/export functions facilitate the interchange of audio files with other desktop system using WAV and GSM audio files * easy to export IVR tree file and its audio files in only on ZIP file and can import this ZIP file * easy to change the nodes, links colors when design trees or in simulator * easy to edit the properties of each node through the properties panel * easy to create/delete any variable and change its value through the variables panel
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SIP SDK 3.0
SIP SDK for .NET and ActiveX - 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 and websites. It accelerates the development of SIP/ RTP compliant soft phone with a fully-customizable user interface and brand name. The conaito 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 suppression, noise suppression, reverb suppression and mute microphon Key Features * 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, Speex, GSM6.10, iLBC, L16, g723 and g729 Codec) * Encrypt SIP account settings (protect your SIP account settings in websites) * Secure Weblicensing (protect your license in websites) * Multi-User conference support * Multi-line (simultaneous calls) support (Multiple Concurrent calls) * Call Hold support * Call Transfer support * Instant text messaging (MIME) support and typing indication * Mute microphone/speaker for each line * DNS SRV resolution for SIP servers (RFC 3263) * RTCP * Auto-answer * Do Not Disturb (DND) * Adaptive jitter buffer * Adaptive silence * PLC (Packet Lost Concealment) * AGC, VAD, AEC, Noise Concealment ...and much more. Try it today!
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