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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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VoIP H.323 SDK A powerful and highly versatile VoIP SDK to accelerate development of H.323 applications and websites Our brand-new VoIP H.323 SDK provides a powerful and highly versatile solution to add quickly H.323 based dial and receive phone calls features in your software applications. It accelerates the development of H.323 compliant soft phone with a fully-customizable user interface and brand name. The conaito VoIP H.323 SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and H.323 compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either by speaking and delivers superior voice quality. It supports DTMF, adaptive silence detection, adaptive jitter buffer! conaito VoIP H.323 SDK is based on IETF standards (H.323, RTP/RTCP, STUN, ICE, etc.), so it should be compatible with other standard based products such as: Asterisk. Key Features * Easily make and receive H.323 based phone calls through any H323 gateway or H.323 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, LPC-10) * Registration on H.323 Server (H.323 Registrar) * Mute Microphone/ Speaker * Adaptive Jitter Buffer * Adaptive Silence Detection * DTMF tones support * Audio device selection * Fully-customizable user interface * Microsoft Authenticode Certificate * Works with all kind of Internet connections * Royalty free licensing * No Yearly/Monthly fee * Very easy to incorporate * Comes as ActiveX control * Fully commented sample applications for various programming languages * Comes now with C#, C++, Delphi 6.0 * For .NET framework as well and all development environments with ActiveX support Advanced digital voice processing features: * Acoustic echo cancellation or suppression * Noise cancellation or suppression

SIP DLL: http://www.conaito.com/trialdownloads/conaitoVoIP_SIP_SDK.zip

VoIP H.323 SDK for .NET and ActiveX: http://www.conaito.com/trialdownloads/conaitoVoIP_H323_SDK.zip

 

Collanos Phone is a software-only Voice over Internet Protocol (VoIP) phone and a multi-protocol instant messenger combined in one application. The main features include placing audio & video calls and sending instant messages, including transparently to existing Jabber, MSN, ICQ, AOL, and Yahoo! networks. Following Collanos` multi-platform strategy, the product is available in versions running on Windows, Mac OSX, and Linux. All basic features of Collanos Phone are available at no cost. Additional premium features will be rolled out. FEATURE OVERVIEW - Audio and video calls - 3-way conference calls - Instant messenger (IM) - Integration with existing IM networks: AOL, ICQ, MSN, Yahoo! and Jabber - Runs on Windows, Mac OSX and Linux - Based on existing Collanos user profiles (i.e. profile is shared with Collanos Workplace) STANDARDS & PROTOCOLS The phone is based on the popular Session Initiation Protocol (SIP) standard, the worldwide de-facto standard for VoIP solutions. The instant messenger part is based on the popular Extensible Messaging and Presence Protocol (XMPP) standard. It integrates with several existing proprietary IM networks in parallel. LICENSE Collanos Phone is based on many pre-existing open-source components and governed under the General Public License (GPL). GUIDE A short user guide can be found at http://www.collanos.com/help/phone/user_guide 

Collanos Phone Windows: http://www.collanos.com/files/downloads/CollanosPhone/CollanosPhoneInstaller.msi

 

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