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, Speex-wb, GSM6.10, iLBC, L16 and g729 g723 Codec
? Open standards-based and interoperable with all of the major equipment vendors
? UDP and TCP support
? Multi-party voice conference support/ Conference split and join, locally mixed conferences
? Multi-line support (multiple simultaneous calls)
? SIP Instant/Chat Messaging with send/receive controlling
? Integrated STUN, TURN and ICE support
? Comes with new sample SIP Proxy Server to provide in bundle with the conaito SIP Client ActiveX a ready up own SIP VoIP and Instant Messaging network solution.
? P2P support for directly connections between 2 SIP clients without SIP Server
? Outbound proxy server support
? Encrypted SIP account settings (encrypted SIP account settings in your webpage)
? Line Hold/Un-hold support
? Call forwarding and rejection
? Call transfer support
? Select media input/output devices
on-the-fly - also during a conversation/ conference)
? Mute microphone/speaker + level indicator
? Auto-answer
? DND (Do Not Disturb)
? Adaptive Jitter buffer
? PLC (Packet Lost Concealment)
? AGC (auto gain controller)
? AES (Acoustic echo cancellation or suppression)
? Noise cancellation or suppression
? DTMF tones support (generation/detection)
? Recording voice conversation into PCM WAVE (.wav) file
? Playing PCM WAVE (.wav) files to the remote end
? Audio file memory cache
? Extended SIP URL functions
? Dynamically loadable codec support (coming soon)
? Comes as ActiveX control (Web demo with ready-up signed CAB included)
? Registration on SIP Server (SIP Registrar)
? Log file on/off setting
? Microphone and Speaker Volume with Mute support