VoIP SIP Speech to Text Call Recorder

Ozeki VoIP SIP .NET SDK for speech to text based call recording system development. The speech to text based call recorder program ensures easy call documentation and traceability by saving the keywords of a telephone conversation into a database. These saved keywords then can be categorized, this way later trace back and searches will be more efficient. The sample program has basic telephone functions like setup and receive calls, DTMF signal sending and receiving, displaying the event on the interface, keyword searching in audio communication. After running the program the telephone automatically registers to the given SIP PBX with the given SIP account. From this point the softphone is ready to establish and receive calls, to send and receive DTMF signals during calls for navigating in IVR systems. With Ozeki VoIP SIP SDK other advanced telephone functions can be added to the sample program such as call forwarding. The component can be initialized with user define specific number of lines. This means that conference call, dial or receive multiple telephone calls, consult call transfer and other valuable functions can be achieved. Ozeki VoIP SIP SDK is capable to deliver outstanding voice quality which is due to the extended codec support. The ability to deliver this remarkable voice quality is not just for high-bandwith users but for low-bandwith users and SIP compatible devices (hardware and software) as well. Voice quality improvement is secured by G.711 aLaw/uLaw, G.729, iLBC, GSM, G.722, SPEEX, SPEEX-WB. With Ozeki VoIP SIP SDK connection to various PBX systems like Cisco Unified CM, Asterisk PBX, 3CX Phone System or to numerous VoIP service providers over the Internet such as Skype Connect, Gafachi, Callcentric is ensured.