VoIP SIP SDK 3.0

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 voice activity detection (VAD), auto gain controller (AGC), acoustic echo suppression (AES), noise suppression, reverb suppression and mute microphone/speaker for each line. It supports multiple lines, call hold, call transfer, DTMF, adaptive silence detection, adaptive jitter buffer, Packet Lost Concealment (PLC), record MP3, WAV and play it!