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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

 

ManageEngine™ VQManager is a powerful, web-based, passive QoS monitoring tool for VoIP networks. It can monitor any device that supports SIP, H.323, Cisco™ Skinny and RTP / RTCP. VQManager monitors real-time VoIP QoS metrics to give instantaneous reports, self-configurable alerts & in-depth fault probing – all from a single web interface. How does VQManager help keep your VoIP infrastructure up and optimized? - Real-time, 24x7 non-intrusive monitoring of various QoS metrics allows speedy identification of VoIP faults and quality deterioration. - VQManager pro-actively notifies faults by means of operator alerts, Email notifications and SNMP traps. - Complete call level and end-point level 24x7 QoS analysis makes possible in-depth fault probing through call traces and call summaries. - VQManager’s supports monitoring of any SIP, H.323 or Cisco™ Skinny device. Also, by processing Cisco (4, 4.x, 5), Swyx 6.0, Vovida Vocal, Tekelec 5.6, Shoretel 6.1, PortaOne and Asterisk CDRs, VQManager keeps the administrator updated with the happenings in the VoIP network regardless of the devices used. - The tool’s powerful and flexible reporting generates over 30 different reports giving insight into usage trends, bandwidth utilization & capacity planning. VQManager requires no other special hardware/software for support and its remotely accessible, user-friendly interface saves precious time & greatly improves administrator productivity. With minimal to zero deployment, VQManager is your fastest and most cost-effective way of ensuring an optimized VoIP communication infrastructure. NOTE: Downloaded software would have support for monitoring unlimited IP phones for the first 30 days of usage. On completion of 30 days, support is reduced to monitoring of 10 IP phones. 

ManageEngine VQManager: http://www.manageengine.com/products/vqmanager/91408665/ManageEngine_VQManager.exe

 

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