Introduction - conaito VoIP Enterprise SDK

Overview of the Features


We describe below an overview of the features of conaito VoIP Enterprise SDK.



The conaito VoIP Enterprise SDK enables VoIP connections to be made in different ways according to developer and operational requirements. This provides the options of using the conaito VoIP Server SDK either for maintaining user connections or, in addition, as a voice-mixing-engine.


When used to maintain user connections the server keeps a list of the respective user registrations and their availability status (online, absent). In this way it is possible to maintain and coordinate all of the VoIP and/or text messaging connections between all current users.


At the time of user registration, when they log on to the VoIP service, this information can be saved and maintained on the server. At the same time, there is the option to send other information to server and this can be specified by the developer.


The server can manage data routing between participants and this includes voice messages and text messages.


However, in this mode of operation there is no voice stream traffic through the server. With multiple connections voice streams can consume considerable bandwidth. With this option the server is merely keeping the connections alive so as to manage the sensing and receiving of messages. As a result this system provides a considerable bandwidth economy at the server level and this in turn economizes on server hardware requirements. In this solution the client applications (conaito VoIP Client SDK) accommodate and manage all of the traffic.


If the Internet connection is of lower bandwidth than is required by the client application, then the conaito VoIP Server SDK can manage all of the voice streaming.


There is even a third option, a mixed mode, in which clients can be connected direct p2p and via proxy server so as to make possible the capture of a large feasible user base.



An important feature of the conaito VoIP Enterprise SDK is the conference function. This enables the simultaneous communication between several participants. The conaito VoIP server SDK solution has the unusual capability of manageing the coordination of many simultaneous conference connections which can running on the basis of real time p2p or proxy server modes at the same time.


Find out more by clicking here.



An important technical capability is conaito VoIP Enterprise SDK’s support of VAD (Voice Activity Detection). This feature can be enabled by users and with this the conaito VoIP operation initiates voice data only when a user speaks. The sensitivity and the silence duration (TimeOut for sending voice after last detect voice activity on mic) can be set by users. This means no data is transmitted when there is a gap in conversation and results in a considerable drop in the demand for bandwith through the intelligent elimination of redundant data packets.



Conaito VoIP operations, either one-to-one or conferences modes, are enntriely secure. Use is made of Blow Fish cryptography methods to encrypt and decrypt voice and text data. The system’s VoIP components manage the encryption/send and receive/decryption processing automatically over the Internet. This provides very secure conferencing!



Recording of conversations or conferences is another very important feature of conaito VoIP. This enables clear recordings to be taken during a conversation or conference. These are saved as named wave (*.wav) files and can be played back later.



The conaito VoIP Enterprise SDK includes extensive text messaging capabilities for sending and receiving text messages in an integrated way. This capability is not available with either the Standard or the Extended SDK.

With this capability VoIp can be combines with chat and it is possible to broaden the application functionality by introducing other protocols such as file transfer.



There is also a wispering function which permits a single person in a conference to contact another specific individual on a one-to-one basis without the text message being seen by other conference participants. This is useful for advisers to send messages to a manager during the course of a conference. This can be information he has requested or data which can be sent without interrupting talk streams.



The conaito VoIP Enterprise SDK allows users to define their availability status such as “available”, “busy” or “not available”. This assists others to select times to call when convenient to the person receiving the call. It also helps control the number of input connections.



The conaito SDK has a number of additional and integrated functions which are useful for developers. These include the listing of participants on the log in register on the server, plus a range of optional data which can be associated with users, conference participants can be listed, microphones, loud speaker or headset sound controls enable the setting of volume levels, microphone boost (amplification) as well as microphone and headphone mute settings. There are functions to control connection and estimate broadcasted data throughput.



For voice streaming the conaito VoIP SDK uses a direct Full Duplex connection.



By employing the GSM 6.10 codec we achieve an excellent quality of speech on the basis of a very low bandwidth. A single connection for VoIP uses only 1.6 Kbytes/second! Codecs which achieve nearly the same quality of speech require a far higher bandwithch.



Because our systems have relatively low bandwidth demands it is possible for conaito VoIP to work with every internet connection, e.g. modems, ISDN, ADSL, cable modems etc.



The conaito Enterprise SDK integrates “Hole Punching” technology to enable table connections through firewalls or NAT network systems. As a result a user can be on a company network or a simple router and use the VoIP system effectively.


A more detailed explanation of this capability can be accessed by clicking here.



The conaito VoIP SDK makes use of the UDP protocol which involves lower delays in comparison with the common TCP protocol, which is used by other voice streaming protocols, too (RTP, RTCP, SIP, H323 etc.).



Sometimes bad internet connections can create lags, or delays, in voice streams. The conaito VoIP SDK can detect this occurrence and automatically removes the lags. As a result the system provides a more stable voice stream under unstable network situations.



Our SDK has the Microsoft Authenticode Certificate signature. Therefore a developer does not have any additional expense which would normally be necessary with a downloadable client ActiveX (CAB).



The conaito SDK is easy to understand and this facilitates the work of integrating it into an application. In the download area we present many examples of source code.

Our support team stands ready to answer any questions you might have.



Developers who purchase and use conaito SDKs receive the right to royalty free licensing. This allows developers to distribute as many copies of their application free of any charges. Developers only need to acquire a single license.


To find out more about licensing, click here.



Learn more about the conaito "hole punching" technology, which is integrated

in our conaito VoIP Enterprise SDK.


Our conaito Support Team gladly answers your questions.




Any Questions?


Our Support Forum or Knowledgebase/ Helpdesk is the best way for your questions. By this way can we help many user to find a answer on their questions.

Please feel free to contact the conaito Support Team incase of any question.

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