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