Results 1 - 15 of 80 for voip lib. (0.005 seconds)
-
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. -
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. -
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. -
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. -
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. -
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 -
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. -
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.
-
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. -
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. -
VoIP Monitor enables you to measure and track performance of voice quality across WAN links. Leveraging Cisco® IP SLAs, VoIP Monitor collects and analyzes VoIP performance statistics including MOS, jitter, network latency, packet loss and other important quality of service (QoS) metrics. These features enable you to proactively find the root cause of VoIP performance degradation and measure expected voice quality in advance of a VoIP deployment. -
Embedded Windows CE VOIP Developers Kit is your embedded telephony solution for developing software/hardware to enable device to device phone calls using SIP/H323 phone right at the electronics level. The design is based on Hybrid Combination of RFC SIP Protocol and Windows RTC and is today successfully ported to Windows CE/Pocket PC/Smart Phone. -
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. -
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. -
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.