conaito VoIP SIP SDK 2.2.1

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 auto gain controller, acoustic echo suppression, 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, record and playing WAV!

conaito VoIP SIP SDK is based on IETF standards (SIP, RTP/RTCP, STUN, ICE, etc.), so it should be compatible with other standard based products such as: SER, OpenSER and Asterisk


Here is a list of the main features of the conaito VoIP SIP SDK:


• 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, Speex, GSM6.10, iLBC, L16, g723, g729 Codec)
NEW in v2.1! Added Domain property to correctly handle sip calls
NEW in v2.1! Updated samples