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conaito VoIP SDK ActiveX for developers of VoIP audio applications and webpages, such as voice chat, conference, VoIP, real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user and broadcast.
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conaito VoIP SDK ActiveX for developers of VoIP audio applications and webpages, such as voice chat, conference, VoIP, real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user and broadcast.
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conaito VoIP SDK ActiveX for developers of VoIP audio applications and webpages, such as voice chat, conference, VoIP, real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user and broadcast.
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conaito VoIP SDK ActiveX for developers of VoIP audio applications and webpages, such as voice chat, conference, VoIP, real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user and broadcast.
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conaito VoIP ActiveX library for developers of VoIP audio applications, such as voice chat, conference, VoIP, providing real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user and broadcast
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Vax Voice activeX is the easiest way to add PC-to-PC voice conferencing and text chat over the LAN, WAN or Internet in your applications and Web pages. Using Vax Voice activeX three or more persons can have a real time voice conference and during the conference, they can even record the conversation into wave (.wav) file for later play back.
It also provides the feature to send/receive encrypted voice and text data over the Internet.
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The VoIP DLL, OCX/ActiveX, COM, and NET contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users. conaito VoIP EVO users are arranged in a tree-structure where each node is a room/channel where users can talk, send instant messages and share files. More than 40 users can be in each room/channel and participate in a conference.
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VoIP 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. It accelerates the development of SIP/ RTP compliant soft phone with a fully-customizable user interface and brand name.
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VoIP SIP SDK with DLL, ActiveX and .NET - 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.
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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.
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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.
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Designed for networking and telecom engineers, admins and communication technology educators and students, this VOIP technology quick guide covers all related technologies: SIP, H.323 and xGCP signaling architecture and technologies, VOIP transport and QoS technologies, and video and audio codecs.
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The conaito VoIP EVO SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users. conaito VoIP EVO users are arranged in a tree-structure where each node is a room/channel where users can talk, send instant messages and share files. More than 40 users can be in each room/channel and participate in a conference.
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The conaito VoIP EVO SDK for Pocket PC and Windows Mobile contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users. conaito VoIP EVO users are arranged in a tree-structure where each node is a room/channel where users can talk, send instant messages and share files. More than 40 users can be in each room/channel and participate in a conference.
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SIP Phone DLL allows to make PC-PC, PC-phone, or phone-phone calls or create Instant Messaging (IM) sessions over the Internet VOIP SIP Phone SDK brings protocol support for ActiveX. With this SDK one can create a simple program to connect and start speaking with anyone with a direct IP Address (provided no NAT Router or Firewall is set) or use the Gateway or the Gatekeeper to connect to /or from PSTN lines.